From prathibhab.tvm at gmail.com Mon Apr 1 03:34:43 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 1 Apr 2024 09:04:43 +0530 Subject: [OpenSIPS-Users] dropped calls Message-ID: How to identify the dropped calls in opensips configuration script? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Mon Apr 1 06:06:04 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Mon, 1 Apr 2024 11:36:04 +0530 Subject: [OpenSIPS-Users] REINVITE IN OPENSIPS Message-ID: Hi All, I have a requirement where I need to send ReInvite from opensips after receiving 200 OK. So First I need to confirm whether Opensips is able to send ReInvite ?? I have gone through module b2b_logic and b2b_entities for the same and it seems opensips sends ReInvite for late SDP negotiations but I want to send after successful bridge for some other task. Can anyone suggest to me which module I need to spend my time on to meet my requirements ?? Regards, *Parthesh Bhavsar | Software Engineer | VOIP* *+91 9638867145* *Ecosmob Technologies Pvt Ltd.* -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 1 06:34:26 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 1 Apr 2024 12:04:26 +0530 Subject: [OpenSIPS-Users] Send UDP packets Message-ID: which module and function to use to send UDP packets to external application from opensips? I -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 1 06:35:02 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 1 Apr 2024 12:05:02 +0530 Subject: [OpenSIPS-Users] Send UDP packets In-Reply-To: References: Message-ID: I've used send() function but the data is transmitted as SIP not as UDP. On Mon, 1 Apr 2024 at 12:04, Prathibha B wrote: > which module and function to use to send UDP packets to external > application from opensips? > > I > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 1 08:26:22 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 1 Apr 2024 11:26:22 +0300 Subject: [OpenSIPS-Users] how to debug many dialogs stuck in state 5? In-Reply-To: References: Message-ID: <4de7d18a-9ac5-4b2e-b6df-33d438ec2cb2@opensips.org> You should never use both dialog and b2b modules in the same time, for the same calls. Trying to have OpenSIPS both Proxy and B2B is a clear recipe for disaster. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 30.03.2024 07:53, Babak Yakhchali wrote: > sorry for the late reply. After disabling modules and simplifying > script logic I found that the issue was b2b entities module being > enabled, downgrading to 3.2 solved the problem > > On Wed, Mar 20, 2024 at 5:19 PM Bogdan-Andrei Iancu > wrote: > > Hi, > > What OpenSIPS version are you using? and what module do you use on > top > of the dialog module? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 13.03.2024 13:55, Babak Yakhchali wrote: > > Hi > > When calling and immediately cancelling, the call is ended but > > dlg_list shows the dialog stuck in state 5. Increasing log level > to 4 > > shows these messages: > > Mar 13 15:15:37 : DBG:tm:timer_routine: timer > > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26 > > Mar 13 15:15:37 : DBG:tm:delete_handler: removing 0x7f1c861d0630 > > Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: > > can't delete -- still reffed (-1) > > Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2 > > Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]: > 0x7f1c861d06e0 (28) > > Mar 13 15:15:37 : DBG:tm:delete_handler: done > > Mar 13 15:15:39 : DBG:tm:timer_routine: timer > > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28 > > Mar 13 15:15:39 : DBG:tm:delete_handler: removing 0x7f1c861d0630 > > Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: > > can't delete -- still reffed (-1) > > Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2 > > Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]: > 0x7f1c861d06e0 (30) > > Mar 13 15:15:39 : DBG:tm:delete_handler: done > > > > How can I debug the issue? What are the possible causes of this? > > thanks > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 1 09:19:44 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 1 Apr 2024 12:19:44 +0300 Subject: [OpenSIPS-Users] REINVITE IN OPENSIPS In-Reply-To: References: Message-ID: <1452cb7b-7320-4d2d-93d2-d21092555415@opensips.org> Hi, Check this dlg_send_requential() [1] function. [1] https://opensips.org/html/docs/modules/3.4.x/dialog.html#func_dlg_send_sequential Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 01.04.2024 09:06, Parthesh Bhavsar via Users wrote: > Hi All, > I have a requirement where I need to send ReInvite from opensips after > receiving 200 OK. So First I need to confirm whether Opensips is able > to send ReInvite ?? I have gone through module b2b_logic and > b2b_entities for the same and it seems opensips sends ReInvite for > late SDP negotiations but I want to send after successful bridge for > some other task. > > Can anyone suggest to me which module I need to spend my time on to > meet my requirements ?? > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > *+91 9638867145* > > *Ecosmob Technologies Pvt Ltd.* > > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our > website, any views or opinions presented in this email are solely > those of the originator and do not necessarily represent those of the > Company or its sister concerns. Any liability (in negligence, contract > or otherwise) arising from any third party taking any action, or > refraining from taking any action on the basis of any of the > information contained in this email is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for > the use of the addressee(s) and contains information that is > PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, > distribution, or copying of this communication is prohibited. Please > inform originator if you have received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of > viruses, trojans, similar or new contaminants/malware, interceptions > or interference, and may not be compatible with your systems. You > shall carry out virus/malware scanning on your own before opening any > attachment to this e-mail. The sender of this e-mail and Company > including its sister concerns shall not be liable for any damage that > may incur to you as a result of viruses, incompleteness of this > message, a delay in receipt of this message or any other computer > problems. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 1 09:23:08 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 1 Apr 2024 12:23:08 +0300 Subject: [OpenSIPS-Users] Opensips 3.2 with TCP connintion and DB operation on a single instance . In-Reply-To: References: Message-ID: <4f783c65-ca0b-49f0-9a45-5825569735de@opensips.org> Hi, If you are referring to the "TCP Connect Issues", yes, it is still valid. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.03.2024 14:07, Sasmita Panda wrote: > Hi All , > > Earlier in the very early stage of using opensips I had faced some > issues while using TCP with DB operation . Opensips has evolved a lot > in the past few years with so many additional features . > > Now , can I use TCP globally with opensips 3.2 . My config will do DB > lookup as well . > > https://www.opensips.org/Documentation/Script-Async-3-2#toc10 > Is this relevant to my scenario or what issue is mentioned here  ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 1 09:28:07 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 1 Apr 2024 12:28:07 +0300 Subject: [OpenSIPS-Users] How to access to a column of a location database In-Reply-To: <0ce77a75-bd01-4b73-85fc-614bf7ce4e3c@opensips.org> References: <89d962bc-0e78-43ef-a5e9-092c11e02b5b@opensips.org> <0ce77a75-bd01-4b73-85fc-614bf7ce4e3c@opensips.org> Message-ID: Hi, With that approach you still have to do the actual lookup(), something that Guillaume wants to avoid, AFAIU. I mean if you do end up doing lookup(), you can check the $socket_out directly :) If you want to fetch the socket info WITHOUT doing lookup() (and affecting the current SIP message), you can do this via the MI `ul_show_contct` [1] directly via script using the mi_script module [2] [1] https://opensips.org/html/docs/modules/3.4.x/usrloc.html#mi_ul_show_contact [2] https://opensips.org/html/docs/modules/3.4.x/mi_script.html#func_mi Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.03.2024 16:01, Vlad Paiu wrote: > > Hello, > > You can use > https://opensips.org/html/docs/modules/3.3.x/registrar.html#param_attr_avp > in order to populate any custom info at save() time and the info will > automatically be populated for you at lookup() time. > > > Regards, > Vlad > > > On 29.03.2024 15:08, guillaume.desgeorge at orange.com wrote: >> >> Hi Bogdan, >> >> Thank you for your answer. >> >> And is it possible to use that lookup("location") function to put the >> “registered socket” field in a variable in order to use it for my >> script ? >> >> Regards, >> >>   Guillaume >> >> *De :*Bogdan-Andrei Iancu >> *Envoyé :* jeudi 28 mars 2024 11:15 >> *À :* OpenSIPS users mailling list ; >> DESGEORGE Guillaume INNOV/IT-S >> *Objet :* Re: [OpenSIPS-Users] How to access to a column of a >> location database >> >> *CAUTION*: This email originated outside the company. Do not click on >> any links or open attachments unless you are expecting them from the >> sender. >> >> *ATTENTION*: Cet e-mail provient de l'extérieur de l'entreprise. Ne >> cliquez pas sur les liens ou n'ouvrez pas les pièces jointes à moins >> de connaitre l'expéditeur. >> >> >> >> Hi Guillaume, >> >> The registered contact (and the additional info) is fetched via the >> lookup("location") function. It also sets the registered socket for >> the routing the current request. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 22.03.2024 10:57, guillaume.desgeorge at orange.com wrote: >> >> Hi everybody, >> >> In my routing logic of my “opensips.cfg” file, I’m trying to >> access the “socket” field of the location table of registered >> contacts. >> >> I have the registering informations correctly written in MySQL >> location table but can’t find a function to access it. >> >> I’d like to have a function which I give the registered username >> and can give me back the associated socket. >> >> Is the lookup () function the good one ? I didn’t understand how >> to use it that way. >> >> Thanks for your help, >> >>    Guillaume >> >> Orange Restricted >> >> Orange Restricted >> >> ____________________________________________________________________________________________________________ >> >> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc >> >> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler >> >> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, >> >> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. >> >> >> >> This message and its attachments may contain confidential or privileged information that may be protected by law; >> >> they should not be distributed, used or copied without authorisation. >> >> If you have received this email in error, please notify the sender and delete this message and its attachments. >> >> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. >> >> Thank you. >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> ____________________________________________________________________________________________________________ >> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc >> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler >> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, >> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. >> >> This message and its attachments may contain confidential or privileged information that may be protected by law; >> they should not be distributed, used or copied without authorisation. >> If you have received this email in error, please notify the sender and delete this message and its attachments. >> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. >> Thank you. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 1 09:39:44 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 1 Apr 2024 12:39:44 +0300 Subject: [OpenSIPS-Users] Why does tracer uri=file show in/out sip messages and uri=sip only shows in? In-Reply-To: <7fb4ccf1-b808-4a24-b403-3f9e52c8c63a@schu.net> References: <312a3ea1-c25d-45a0-aea2-10bd0cecc090@schu.net> <7fb4ccf1-b808-4a24-b403-3f9e52c8c63a@schu.net> Message-ID: Hi Matthew, If I understand correctly, if you use EITHER sip, EITHER file tracing (but only one at the time), it works ok. But using both in the same time doesn't ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.03.2024 19:57, Matthew Schumacher wrote: > On 3/29/24 9:08 AM, Matthew Schumacher wrote: >> Hello All, >> >> I have this config: >> >> modparam("tracer","trace_id","[siptrace]uri=sip:127.0.0.1:9999") >> modparam("tracer","trace_id","[siptrace]uri=file:/var/log/siptrace.log") >> >> route { >>     trace("siptrace","d","sip"); >> >> ... >> >> When I look at the messages that show up on UDP:9999 I only see >> egress sip messages, but when I look at /var/log/siptrace.log I see >> both sides of the conversation.  Is this by design?  Is there a way >> to send all messages to a port, or a socket, or perhaps define a pipe >> to another process?  I'm trying to setup a second daemon that >> monitors all of the sip dialog between opensips and other hosts. >> >> Thanks! >> Matt >> > > > Correction, I only see ingress SIP messages, not egress when using the > sip: uri in the tracer module.  If I use file: or hep: I see both sides. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kwem at gmx.de Mon Apr 1 13:58:38 2024 From: kwem at gmx.de (Karsten Wemheuer) Date: Mon, 01 Apr 2024 15:58:38 +0200 Subject: [OpenSIPS-Users] B2BUA Broken ACKs In-Reply-To: References: Message-ID: Hi, unfortunately I can't look at the image, as it is embedded in the text. I would check the via header of the SIP messages causing the ACK. Maybe OpenSIPS just doesn't know where to send the ACK to. Regards, Karsten  Am Sonntag, dem 31.03.2024 um 00:56 +0500 schrieb Naveed Ur Rehman: > image.png > Hi All > I have a strange issue, where ACKs are not reaching the called party. > Even BYE messages reach back to called party fine, but ACKs are > stucked within opensips,. opensips not even trying to send the ACKs > to callee. What could be the problem? > > Regards > > Naveed > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From parthesh.bhavsar at ecosmob.com Tue Apr 2 06:26:25 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Tue, 2 Apr 2024 11:56:25 +0530 Subject: [OpenSIPS-Users] REINVITE IN OPENSIPS In-Reply-To: <1452cb7b-7320-4d2d-93d2-d21092555415@opensips.org> References: <1452cb7b-7320-4d2d-93d2-d21092555415@opensips.org> Message-ID: Thanks, It works!!! Regards, *Parthesh Bhavsar | Software Engineer | VOIP* On Mon, Apr 1, 2024 at 2:50 PM Bogdan-Andrei Iancu wrote: > Hi, > > Check this dlg_send_requential() [1] function. > > [1] > https://opensips.org/html/docs/modules/3.4.x/dialog.html#func_dlg_send_sequential > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 01.04.2024 09:06, Parthesh Bhavsar via Users wrote: > > Hi All, > I have a requirement where I need to send ReInvite from opensips after > receiving 200 OK. So First I need to confirm whether Opensips is able to > send ReInvite ?? I have gone through module b2b_logic and b2b_entities > for the same and it seems opensips sends ReInvite for late SDP negotiations > but I want to send after successful bridge for some other task. > > Can anyone suggest to me which module I need to spend my time on to meet > my requirements ?? > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > *+91 9638867145* > > *Ecosmob Technologies Pvt Ltd.* > > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 06:36:24 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 12:06:24 +0530 Subject: [OpenSIPS-Users] unix timestamp in microseconds Message-ID: How to get the unix timestamp in microseconds? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From guillaume.desgeorge at orange.com Tue Apr 2 09:05:41 2024 From: guillaume.desgeorge at orange.com (guillaume.desgeorge at orange.com) Date: Tue, 2 Apr 2024 09:05:41 +0000 Subject: [OpenSIPS-Users] How to access to a column of a location database In-Reply-To: References: <89d962bc-0e78-43ef-a5e9-092c11e02b5b@opensips.org> <0ce77a75-bd01-4b73-85fc-614bf7ce4e3c@opensips.org> Message-ID: Hello, That’s what I needed. Thank you for your help 😊 Regards, Guillaume De : Bogdan-Andrei Iancu Envoyé : lundi 1 avril 2024 11:28 À : OpenSIPS users mailling list ; Vlad Paiu ; DESGEORGE Guillaume INNOV/IT-S Objet : Re: [OpenSIPS-Users] How to access to a column of a location database CAUTION : This email originated outside the company. Do not click on any links or open attachments unless you are expecting them from the sender. ATTENTION : Cet e-mail provient de l'extérieur de l'entreprise. Ne cliquez pas sur les liens ou n'ouvrez pas les pièces jointes à moins de connaitre l'expéditeur. Hi, With that approach you still have to do the actual lookup(), something that Guillaume wants to avoid, AFAIU. I mean if you do end up doing lookup(), you can check the $socket_out directly :) If you want to fetch the socket info WITHOUT doing lookup() (and affecting the current SIP message), you can do this via the MI `ul_show_contct` [1] directly via script using the mi_script module [2] [1] https://opensips.org/html/docs/modules/3.4.x/usrloc.html#mi_ul_show_contact [2] https://opensips.org/html/docs/modules/3.4.x/mi_script.html#func_mi Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.03.2024 16:01, Vlad Paiu wrote: Hello, You can use https://opensips.org/html/docs/modules/3.3.x/registrar.html#param_attr_avp in order to populate any custom info at save() time and the info will automatically be populated for you at lookup() time. Regards, Vlad On 29.03.2024 15:08, guillaume.desgeorge at orange.com wrote: Hi Bogdan, Thank you for your answer. And is it possible to use that lookup("location") function to put the “registered socket” field in a variable in order to use it for my script ? Regards, Guillaume De : Bogdan-Andrei Iancu Envoyé : jeudi 28 mars 2024 11:15 À : OpenSIPS users mailling list ; DESGEORGE Guillaume INNOV/IT-S Objet : Re: [OpenSIPS-Users] How to access to a column of a location database CAUTION : This email originated outside the company. Do not click on any links or open attachments unless you are expecting them from the sender. ATTENTION : Cet e-mail provient de l'extérieur de l'entreprise. Ne cliquez pas sur les liens ou n'ouvrez pas les pièces jointes à moins de connaitre l'expéditeur. Hi Guillaume, The registered contact (and the additional info) is fetched via the lookup("location") function. It also sets the registered socket for the routing the current request. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 22.03.2024 10:57, guillaume.desgeorge at orange.com wrote: Hi everybody, In my routing logic of my “opensips.cfg” file, I’m trying to access the “socket” field of the location table of registered contacts. I have the registering informations correctly written in MySQL location table but can’t find a function to access it. I’d like to have a function which I give the registered username and can give me back the associated socket. Is the lookup () function the good one ? I didn’t understand how to use it that way. Thanks for your help, Guillaume Orange Restricted Orange Restricted ____________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 09:15:38 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 14:45:38 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > You can also use the REST client. And there are many other ways, as well. > > > > There is no single correct answer to the vague question of connecting to > any generic “external application”. You must understand your systems and > decide the best approach depending on the needs and capabilities of both > the external application and OpenSIPS. > > > > Ben Newlin > > > > *From: *Users on behalf of > Bogdan-Andrei Iancu > *Date: *Wednesday, March 20, 2024 at 10:06 AM > *To: *OpenSIPS users mailling list , Prathibha > B > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Use the dialog events: > > https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED > > And you subscribe from outside OpenSIPS for such events: > https://www.opensips.org/Documentation/Interface-Events-3-4 > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 20.03.2024 12:16, Prathibha B wrote: > > No. I want to pass START, CONNECT, END messages from OpenSIPS to external > application. > > > > On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: > > Well, to execute external command from opensips you may want to use EXEC > module. > > this is a manual for v3.2: > > https://opensips.org/html/docs/modules/3.2.x/exec.html > > > > On 3/20/2024 5:00 AM, Prathibha B wrote: > > How to integrate OpenSIPS with external applications? > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Best Regards: > > Marcin Groszek > > Business Phone Service > > https://www.voipplus.net > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 09:21:49 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 14:51:49 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > How to use E_DLG_STATE_CHANGED to identify the start of the call? > > > On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > >> You can also use the REST client. And there are many other ways, as well. >> >> >> >> There is no single correct answer to the vague question of connecting to >> any generic “external application”. You must understand your systems and >> decide the best approach depending on the needs and capabilities of both >> the external application and OpenSIPS. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of >> Bogdan-Andrei Iancu >> *Date: *Wednesday, March 20, 2024 at 10:06 AM >> *To: *OpenSIPS users mailling list , Prathibha >> B >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Use the dialog events: >> >> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >> >> And you subscribe from outside OpenSIPS for such events: >> https://www.opensips.org/Documentation/Interface-Events-3-4 >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> https://www.opensips-solutions.com >> >> https://www.siphub.com >> >> On 20.03.2024 12:16, Prathibha B wrote: >> >> No. I want to pass START, CONNECT, END messages from OpenSIPS to external >> application. >> >> >> >> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: >> >> Well, to execute external command from opensips you may want to use EXEC >> module. >> >> this is a manual for v3.2: >> >> https://opensips.org/html/docs/modules/3.2.x/exec.html >> >> >> >> On 3/20/2024 5:00 AM, Prathibha B wrote: >> >> How to integrate OpenSIPS with external applications? >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> >> Best Regards: >> >> Marcin Groszek >> >> Business Phone Service >> >> https://www.voipplus.net >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 09:29:25 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 14:59:25 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > I tried > event_route[E_DLG_STATE_CHANGED] { > > } > > I am getting syntax error. > > On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > >> How to use E_DLG_STATE_CHANGED to identify the start of the call? >> >> >> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >> >>> You can also use the REST client. And there are many other ways, as well. >>> >>> >>> >>> There is no single correct answer to the vague question of connecting to >>> any generic “external application”. You must understand your systems and >>> decide the best approach depending on the needs and capabilities of both >>> the external application and OpenSIPS. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of >>> Bogdan-Andrei Iancu >>> *Date: *Wednesday, March 20, 2024 at 10:06 AM >>> *To: *OpenSIPS users mailling list , >>> Prathibha B >>> *Subject: *Re: [OpenSIPS-Users] external applications >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Use the dialog events: >>> >>> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >>> >>> And you subscribe from outside OpenSIPS for such events: >>> https://www.opensips.org/Documentation/Interface-Events-3-4 >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> >>> >>> OpenSIPS Founder and Developer >>> >>> https://www.opensips-solutions.com >>> >>> https://www.siphub.com >>> >>> On 20.03.2024 12:16, Prathibha B wrote: >>> >>> No. I want to pass START, CONNECT, END messages from OpenSIPS to >>> external application. >>> >>> >>> >>> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek >>> wrote: >>> >>> Well, to execute external command from opensips you may want to use EXEC >>> module. >>> >>> this is a manual for v3.2: >>> >>> https://opensips.org/html/docs/modules/3.2.x/exec.html >>> >>> >>> >>> On 3/20/2024 5:00 AM, Prathibha B wrote: >>> >>> How to integrate OpenSIPS with external applications? >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> _______________________________________________ >>> >>> Users mailing list >>> >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> >>> Best Regards: >>> >>> Marcin Groszek >>> >>> Business Phone Service >>> >>> https://www.voipplus.net >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> _______________________________________________ >>> >>> Users mailing list >>> >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 09:39:37 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 15:09:37 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > How do I identify the START and TRYING state of the call? > > I am able to capture RINGING, ANSWER and TERMINATED states. > > On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > >> I tried >> event_route[E_DLG_STATE_CHANGED] { >> >> } >> >> I am getting syntax error. >> >> On Tue, 2 Apr 2024 at 14:45, Prathibha B >> wrote: >> >>> How to use E_DLG_STATE_CHANGED to identify the start of the call? >>> >>> >>> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >>> >>>> You can also use the REST client. And there are many other ways, as >>>> well. >>>> >>>> >>>> >>>> There is no single correct answer to the vague question of connecting >>>> to any generic “external application”. You must understand your systems and >>>> decide the best approach depending on the needs and capabilities of both >>>> the external application and OpenSIPS. >>>> >>>> >>>> >>>> Ben Newlin >>>> >>>> >>>> >>>> *From: *Users on behalf of >>>> Bogdan-Andrei Iancu >>>> *Date: *Wednesday, March 20, 2024 at 10:06 AM >>>> *To: *OpenSIPS users mailling list , >>>> Prathibha B >>>> *Subject: *Re: [OpenSIPS-Users] external applications >>>> >>>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>>> >>>> >>>> ------------------------------ >>>> >>>> Use the dialog events: >>>> >>>> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >>>> >>>> And you subscribe from outside OpenSIPS for such events: >>>> https://www.opensips.org/Documentation/Interface-Events-3-4 >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> >>>> >>>> OpenSIPS Founder and Developer >>>> >>>> https://www.opensips-solutions.com >>>> >>>> https://www.siphub.com >>>> >>>> On 20.03.2024 12:16, Prathibha B wrote: >>>> >>>> No. I want to pass START, CONNECT, END messages from OpenSIPS to >>>> external application. >>>> >>>> >>>> >>>> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek >>>> wrote: >>>> >>>> Well, to execute external command from opensips you may want to use >>>> EXEC module. >>>> >>>> this is a manual for v3.2: >>>> >>>> https://opensips.org/html/docs/modules/3.2.x/exec.html >>>> >>>> >>>> >>>> On 3/20/2024 5:00 AM, Prathibha B wrote: >>>> >>>> How to integrate OpenSIPS with external applications? >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> Users mailing list >>>> >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> -- >>>> >>>> Best Regards: >>>> >>>> Marcin Groszek >>>> >>>> Business Phone Service >>>> >>>> https://www.voipplus.net >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> Users mailing list >>>> >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Apr 2 11:18:31 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 2 Apr 2024 16:48:31 +0530 Subject: [OpenSIPS-Users] Trncated SIP header . Message-ID: I am using opensips 3.2 . I have added a custom header in the sip requests like REGISTER and INVITE called X-Tag . This tag (X-Tag header) has a large number of key value pairs that get printed in the logs . Around 50 key value pair strings . I am not able to port the data here as that makes my message very large and that gets rejected before posting in the group . In the wireshark trace in the server the header value gets truncated and also in the DB attr column shows the truncated value . Is this because the attr column size is ( string of 255 ) . If so then can I increase this value or is there any restriction by opensips ? Even if the data is not showing in the DB , is opensips able to save the entire data and process that during the call ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 2 12:04:26 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 2 Apr 2024 17:34:26 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B wrote: > I am able to capture the trying status also. But not getting the START of > the call... > > On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > >> How do I identify the START and TRYING state of the call? >> >> I am able to capture RINGING, ANSWER and TERMINATED states. >> >> On Tue, 2 Apr 2024 at 14:51, Prathibha B >> wrote: >> >>> I tried >>> event_route[E_DLG_STATE_CHANGED] { >>> >>> } >>> >>> I am getting syntax error. >>> >>> On Tue, 2 Apr 2024 at 14:45, Prathibha B >>> wrote: >>> >>>> How to use E_DLG_STATE_CHANGED to identify the start of the call? >>>> >>>> >>>> On Wed, 20 Mar 2024 at 19:46, Ben Newlin >>>> wrote: >>>> >>>>> You can also use the REST client. And there are many other ways, as >>>>> well. >>>>> >>>>> >>>>> >>>>> There is no single correct answer to the vague question of connecting >>>>> to any generic “external application”. You must understand your systems and >>>>> decide the best approach depending on the needs and capabilities of both >>>>> the external application and OpenSIPS. >>>>> >>>>> >>>>> >>>>> Ben Newlin >>>>> >>>>> >>>>> >>>>> *From: *Users on behalf of >>>>> Bogdan-Andrei Iancu >>>>> *Date: *Wednesday, March 20, 2024 at 10:06 AM >>>>> *To: *OpenSIPS users mailling list , >>>>> Prathibha B >>>>> *Subject: *Re: [OpenSIPS-Users] external applications >>>>> >>>>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> Use the dialog events: >>>>> >>>>> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >>>>> >>>>> And you subscribe from outside OpenSIPS for such events: >>>>> https://www.opensips.org/Documentation/Interface-Events-3-4 >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> >>>>> >>>>> OpenSIPS Founder and Developer >>>>> >>>>> https://www.opensips-solutions.com >>>>> >>>>> https://www.siphub.com >>>>> >>>>> On 20.03.2024 12:16, Prathibha B wrote: >>>>> >>>>> No. I want to pass START, CONNECT, END messages from OpenSIPS to >>>>> external application. >>>>> >>>>> >>>>> >>>>> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek >>>>> wrote: >>>>> >>>>> Well, to execute external command from opensips you may want to use >>>>> EXEC module. >>>>> >>>>> this is a manual for v3.2: >>>>> >>>>> https://opensips.org/html/docs/modules/3.2.x/exec.html >>>>> >>>>> >>>>> >>>>> On 3/20/2024 5:00 AM, Prathibha B wrote: >>>>> >>>>> How to integrate OpenSIPS with external applications? >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Regards, >>>>> >>>>> B.Prathibha >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> >>>>> Users mailing list >>>>> >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> -- >>>>> >>>>> Best Regards: >>>>> >>>>> Marcin Groszek >>>>> >>>>> Business Phone Service >>>>> >>>>> https://www.voipplus.net >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Regards, >>>>> >>>>> B.Prathibha >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> >>>>> Users mailing list >>>>> >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From medeanwz at gmail.com Tue Apr 2 12:57:30 2024 From: medeanwz at gmail.com (M S) Date: Tue, 2 Apr 2024 14:57:30 +0200 Subject: [OpenSIPS-Users] how to debug many dialogs stuck in state 5? In-Reply-To: <4de7d18a-9ac5-4b2e-b6df-33d438ec2cb2@opensips.org> References: <4de7d18a-9ac5-4b2e-b6df-33d438ec2cb2@opensips.org> Message-ID: Hi Bogdan, When you say at the same time, you mean in the same script? or same route? or same block for example? Also what about using dialog and modules that use b2b in underlying layers, for example mediaexchange? Thank you! On Mon, Apr 1, 2024 at 10:28 AM Bogdan-Andrei Iancu wrote: > You should never use both dialog and b2b modules in the same time, for the > same calls. Trying to have OpenSIPS both Proxy and B2B is a clear recipe > for disaster. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 30.03.2024 07:53, Babak Yakhchali wrote: > > sorry for the late reply. After disabling modules and simplifying script > logic I found that the issue was b2b entities module being enabled, > downgrading to 3.2 solved the problem > > On Wed, Mar 20, 2024 at 5:19 PM Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> What OpenSIPS version are you using? and what module do you use on top >> of the dialog module? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 13.03.2024 13:55, Babak Yakhchali wrote: >> > Hi >> > When calling and immediately cancelling, the call is ended but >> > dlg_list shows the dialog stuck in state 5. Increasing log level to 4 >> > shows these messages: >> > Mar 13 15:15:37 : DBG:tm:timer_routine: timer >> > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26 >> > Mar 13 15:15:37 : DBG:tm:delete_handler: removing 0x7f1c861d0630 >> > Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: >> > can't delete -- still reffed (-1) >> > Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2 >> > Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]: 0x7f1c861d06e0 (28) >> > Mar 13 15:15:37 : DBG:tm:delete_handler: done >> > Mar 13 15:15:39 : DBG:tm:timer_routine: timer >> > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28 >> > Mar 13 15:15:39 : DBG:tm:delete_handler: removing 0x7f1c861d0630 >> > Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: >> > can't delete -- still reffed (-1) >> > Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2 >> > Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]: 0x7f1c861d06e0 (30) >> > Mar 13 15:15:39 : DBG:tm:delete_handler: done >> > >> > How can I debug the issue? What are the possible causes of this? >> > thanks >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 2 14:56:22 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 2 Apr 2024 14:56:22 +0000 Subject: [OpenSIPS-Users] unix timestamp in microseconds In-Reply-To: References: Message-ID: There are many ways and they are all clearly documented: https://opensips.org/docs/modules/3.4.x/cfgutils.html#func_get_accurate_time https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#get_timestamp https://www.opensips.org/Documentation/Script-CoreVar-3-4#time https://www.opensips.org/Documentation/Script-CoreVar-3-4#Tsm Please do try to find answers on your own before engaging the mailing list with such trivial questions. Ben Newlin From: Users on behalf of Prathibha B Date: Tuesday, April 2, 2024 at 2:37 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] unix timestamp in microseconds EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ How to get the unix timestamp in microseconds? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 2 15:01:56 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 2 Apr 2024 15:01:56 +0000 Subject: [OpenSIPS-Users] Trncated SIP header . In-Reply-To: References: Message-ID: We use PostgreSQL and we ran into this as well. We altered all of our tables to use type “TEXT” for these columns, which has no length limit, instead of the default VARCHAR(255). We have not had any issues with this; OpenSIPS doesn’t seem to care about the length. Ben Newlin From: Users on behalf of Sasmita Panda Date: Tuesday, April 2, 2024 at 7:20 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Trncated SIP header . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am using opensips 3.2 . I have added a custom header in the sip requests like REGISTER and INVITE called X-Tag . This tag (X-Tag header) has a large number of key value pairs that get printed in the logs . Around 50 key value pair strings . I am not able to port the data here as that makes my message very large and that gets rejected before posting in the group . In the wireshark trace in the server the header value gets truncated and also in the DB attr column shows the truncated value . Is this because the attr column size is ( string of 255 ) . If so then can I increase this value or is there any restriction by opensips ? Even if the data is not showing in the DB , is opensips able to save the entire data and process that during the call ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 2 15:05:00 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 2 Apr 2024 15:05:00 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users on behalf of Prathibha B Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 04:52:10 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 10:22:10 +0530 Subject: [OpenSIPS-Users] Trying for more than 60 secs Message-ID: modparam("tm", "fr_timeout", 45) modparam("tm", "fr_inv_timeout", 60) These are the tm parameters I've set in opensips. When User B is not available, User keeps on Trying without stopping. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Apr 3 05:28:43 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 3 Apr 2024 10:58:43 +0530 Subject: [OpenSIPS-Users] Trncated SIP header . In-Reply-To: References: Message-ID: Thank you Ben . Will try the same . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Apr 2, 2024 at 8:33 PM Ben Newlin wrote: > We use PostgreSQL and we ran into this as well. We altered all of our > tables to use type “TEXT” for these columns, which has no length limit, > instead of the default VARCHAR(255). We have not had any issues with this; > OpenSIPS doesn’t seem to care about the length. > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Tuesday, April 2, 2024 at 7:20 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Trncated SIP header . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I am using opensips 3.2 . I have added a custom header in the sip > requests like REGISTER and INVITE called X-Tag . > > > > This tag (X-Tag header) has a large number of key value pairs that get > printed in the logs . Around 50 key value pair strings . I am not able to > port the data here as that makes my message very large and that gets > rejected before posting in the group . > > > > In the wireshark trace in the server the header value gets truncated and > also in the DB attr column shows the truncated value . Is this because the > attr column size is ( string of 255 ) . If so then can I increase this > value or is there any restriction by opensips ? > > > > Even if the data is not showing in the DB , is opensips able to save the > entire data and process that during the call ? > > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 06:34:32 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 12:04:32 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: > The start of the call would be when you call “create_dialog”. The dialog > state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change > event is raised for creation. It may only be raised when the state changes > after creation. But since you control the dialog creation, you can just > take whatever action you desire at that time. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Tuesday, April 2, 2024 at 8:05 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I tried is_method("INVITE"), but it is getting called only at the start of > RINGING. > > > > On Tue, 2 Apr 2024 at 15:09, Prathibha B wrote: > > I am able to capture the trying status also. But not getting the START of > the call... > > > > On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > > How do I identify the START and TRYING state of the call? > > > > I am able to capture RINGING, ANSWER and TERMINATED states. > > > > On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > > I tried > > event_route[E_DLG_STATE_CHANGED] { > > } > > > > I am getting syntax error. > > > > On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > > How to use *E_DLG_STATE_CHANGED to identify the start of the call?* > > > > > > On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > > You can also use the REST client. And there are many other ways, as well. > > > > There is no single correct answer to the vague question of connecting to > any generic “external application”. You must understand your systems and > decide the best approach depending on the needs and capabilities of both > the external application and OpenSIPS. > > > > Ben Newlin > > > > *From: *Users on behalf of > Bogdan-Andrei Iancu > *Date: *Wednesday, March 20, 2024 at 10:06 AM > *To: *OpenSIPS users mailling list , Prathibha > B > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Use the dialog events: > > https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED > > And you subscribe from outside OpenSIPS for such events: > https://www.opensips.org/Documentation/Interface-Events-3-4 > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 20.03.2024 12:16, Prathibha B wrote: > > No. I want to pass START, CONNECT, END messages from OpenSIPS to external > application. > > > > On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: > > Well, to execute external command from opensips you may want to use EXEC > module. > > this is a manual for v3.2: > > https://opensips.org/html/docs/modules/3.2.x/exec.html > > > > On 3/20/2024 5:00 AM, Prathibha B wrote: > > How to integrate OpenSIPS with external applications? > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Best Regards: > > Marcin Groszek > > Business Phone Service > > https://www.voipplus.net > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 10:32:58 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 16:02:58 +0530 Subject: [OpenSIPS-Users] Event ID Message-ID: Is it possible to retrieve the event ID passed to the client in opensips? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 10:35:02 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 16:05:02 +0530 Subject: [OpenSIPS-Users] Event ID In-Reply-To: References: Message-ID: I have to pass this event ID to external application. On Wed, 3 Apr 2024 at 16:02, Prathibha B wrote: > Is it possible to retrieve the event ID passed to the client in opensips? > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Apr 3 10:54:24 2024 From: nzdealshelp at gmail.com (nz deals) Date: Wed, 3 Apr 2024 23:54:24 +1300 Subject: [OpenSIPS-Users] dashboard stats from opensips Message-ID: Hi everyone, I'm seeking guidance on creating a dashboard. I'm considering saving dialog events in Redis (straight from OpenSIPS), allowing my dashboard to directly access the Redis cache. Do you think this is a wise strategy, or do you have any alternative suggestions? Any expert's suggestion will be highly appreciated. In fact if someone has any example to check, raise dialog events like call, ringing, 183, answered , cancel and bye etc... Thank you Regards, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 12:06:17 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 17:36:17 +0530 Subject: [OpenSIPS-Users] location table Message-ID: Where does opensips store the location details of the end points? I checked in location table in the database. It contains only very few entries and those are very old. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 12:06:47 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 17:36:47 +0530 Subject: [OpenSIPS-Users] unix timestamp in microseconds In-Reply-To: References: Message-ID: ok. On Tue, 2 Apr 2024 at 20:29, Ben Newlin wrote: > There are many ways and they are all clearly documented: > > > > > https://opensips.org/docs/modules/3.4.x/cfgutils.html#func_get_accurate_time > > > https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#get_timestamp > > https://www.opensips.org/Documentation/Script-CoreVar-3-4#time > > https://www.opensips.org/Documentation/Script-CoreVar-3-4#Tsm > > > > Please do try to find answers on your own before engaging the mailing list > with such trivial questions. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Tuesday, April 2, 2024 at 2:37 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] unix timestamp in microseconds > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > How to get the unix timestamp in microseconds? > > > > -- > > Regards, > > B.Prathibha > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 3 13:30:22 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 3 Apr 2024 13:30:22 +0000 Subject: [OpenSIPS-Users] Trying for more than 60 secs In-Reply-To: References: Message-ID: Can you clarify what you mean when you say “User keeps on Trying”? Do you mean the calling party? Those settings only affect OpenSIPS’ behavior. Also confirm what you mean by “User B is not available”. Is the SIP endpoint not responding at all, is it sending provisional responses, or is it sending a “Not Available” response like 480? These settings only control behavior in the first scenario. If User B is sending responses, it will reset the timer with every response, that is per RFC3261 (this was actually discussed in another very recent thread here). Ben Newlin From: Users on behalf of Prathibha B Date: Wednesday, April 3, 2024 at 12:54 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Trying for more than 60 secs EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ modparam("tm", "fr_timeout", 45) modparam("tm", "fr_inv_timeout", 60) These are the tm parameters I've set in opensips. When User B is not available, User keeps on Trying without stopping. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 3 13:33:33 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 3 Apr 2024 13:33:33 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users on behalf of Prathibha B Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Apr 3 13:33:59 2024 From: Johan at democon.be (Johan De Clercq) Date: Wed, 3 Apr 2024 15:33:59 +0200 Subject: [OpenSIPS-Users] memory fragmentation when calling dr_reload. Message-ID: Hi, A client has a very big dynamic routing rule set. (dr_rules >= 2.1 gb ). When reloading the db in opensips (dr_reload), I see below error in the log ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes left), please increase the "-M" command line parameter! the -M parameter is now at 256. Should I increase this to 3000 to work aroun this issue ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Apr 3 14:04:31 2024 From: Johan at democon.be (Johan De Clercq) Date: Wed, 3 Apr 2024 16:04:31 +0200 Subject: [OpenSIPS-Users] memory fragmentation when calling dr_reload. In-Reply-To: References: Message-ID: In addtion, I have 24 children, so can I increase in some way only the process that handles the fifo requests ? Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq : > Hi, > > A client has a very big dynamic routing rule set. (dr_rules >= 2.1 gb ). > When reloading the db in opensips (dr_reload), I see below error in the > log > > ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes left), > please increase the "-M" command line parameter! > > the -M parameter is now at 256. Should I increase this to 3000 to work > aroun this issue ? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 3 14:13:01 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 3 Apr 2024 14:13:01 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: Yes. I am canceling the call prior to dialog creation. Sent from Outlook for Android ________________________________ From: Users on behalf of Ben Newlin Sent: Wednesday, April 3, 2024 7:03:33 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users on behalf of Prathibha B Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 3 14:26:45 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 3 Apr 2024 17:26:45 +0300 Subject: [OpenSIPS-Users] memory fragmentation when calling dr_reload. In-Reply-To: References: Message-ID: What OpenSIPS version is there ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 17:04, Johan De Clercq wrote: > In addtion, I have 24 children, so can I increase in some way only the > process that handles the fifo requests ? > > Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq : > > Hi, > > A client has a very big dynamic routing rule set. (dr_rules >= 2.1 > gb ). > When reloading the db in opensips (dr_reload), I see below error > in the log > > ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes > left), please increase the "-M" command line parameter! > the -M parameter is now at 256. Should I increase this to 3000 to > work aroun this issue ? > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 3 14:27:31 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 3 Apr 2024 17:27:31 +0300 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: Message-ID: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Hi Jason, Have you checked the Dashboard in OpenSIPS Control Panel ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 13:54, nz deals wrote: > Hi everyone, > > I'm seeking guidance on creating a dashboard. I'm considering saving > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > to directly access the Redis cache. Do you think this is a wise > strategy, or do you have any alternative suggestions? Any expert's > suggestion will be highly appreciated. In fact if someone has any > example to check, raise dialog events like call, ringing, 183, > answered , cancel and bye etc... > > Thank you > > Regards, > Jason > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From nzdealshelp at gmail.com Wed Apr 3 17:01:55 2024 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 4 Apr 2024 06:01:55 +1300 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: Hi Bogdan, Yes, something along the lines of the OpenSIPs control panel, but I'm looking for very basic statistics, such as the details of currently active calls and a straightforward graph displaying concurrent calls. Thank you Regards, Jason On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu wrote: > Hi Jason, > > Have you checked the Dashboard in OpenSIPS Control Panel ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 03.04.2024 13:54, nz deals wrote: > > Hi everyone, > > > > I'm seeking guidance on creating a dashboard. I'm considering saving > > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > > to directly access the Redis cache. Do you think this is a wise > > strategy, or do you have any alternative suggestions? Any expert's > > suggestion will be highly appreciated. In fact if someone has any > > example to check, raise dialog events like call, ringing, 183, > > answered , cancel and bye etc... > > > > Thank you > > > > Regards, > > Jason > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 3 17:12:26 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 3 Apr 2024 17:12:26 +0000 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: OpenSIPS will already track some very basic statistics like this for you using the Statistics module. https://opensips.org/docs/modules/3.4.x/statistics.html For example, the Dialog module exposes concurrent calls. We have a scheduled job that queries those stats via MI and pushes them into our external metrics system, allowing us to create dashboards from the data. Ben Newlin From: Users on behalf of nz deals Date: Wednesday, April 3, 2024 at 1:03 PM To: Bogdan-Andrei Iancu Cc: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi Bogdan, Yes, something along the lines of the OpenSIPs control panel, but I'm looking for very basic statistics, such as the details of currently active calls and a straightforward graph displaying concurrent calls. Thank you Regards, Jason On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu > wrote: Hi Jason, Have you checked the Dashboard in OpenSIPS Control Panel ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 13:54, nz deals wrote: > Hi everyone, > > I'm seeking guidance on creating a dashboard. I'm considering saving > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > to directly access the Redis cache. Do you think this is a wise > strategy, or do you have any alternative suggestions? Any expert's > suggestion will be highly appreciated. In fact if someone has any > example to check, raise dialog events like call, ringing, 183, > answered , cancel and bye etc... > > Thank you > > Regards, > Jason > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Apr 3 20:44:05 2024 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 4 Apr 2024 09:44:05 +1300 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: Thanks Ben, The issue with the scheduled task is that it introduces a delay. I'm exploring methods to enable real-time display. By streaming events directly from OpenSIPS, we could achieve live updates on the display. Thank you Regards, Jason On Thu, 4 Apr 2024 at 06:15, Ben Newlin wrote: > OpenSIPS will already track some very basic statistics like this for you > using the Statistics module. > > > > https://opensips.org/docs/modules/3.4.x/statistics.html > > > > For example, the Dialog module exposes concurrent calls. > > > > We have a scheduled job that queries those stats via MI and pushes them > into our external metrics system, allowing us to create dashboards from the > data. > > > > Ben Newlin > > > > *From: *Users on behalf of nz deals < > nzdealshelp at gmail.com> > *Date: *Wednesday, April 3, 2024 at 1:03 PM > *To: *Bogdan-Andrei Iancu > *Cc: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] dashboard stats from opensips > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi Bogdan, > > > > Yes, something along the lines of the OpenSIPs control panel, but I'm > looking for very basic statistics, such as the details of currently active > calls and a straightforward graph displaying concurrent calls. > > > Thank you > > > > Regards, > > Jason > > > > On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu > wrote: > > Hi Jason, > > Have you checked the Dashboard in OpenSIPS Control Panel ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 03.04.2024 13:54, nz deals wrote: > > Hi everyone, > > > > I'm seeking guidance on creating a dashboard. I'm considering saving > > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > > to directly access the Redis cache. Do you think this is a wise > > strategy, or do you have any alternative suggestions? Any expert's > > suggestion will be highly appreciated. In fact if someone has any > > example to check, raise dialog events like call, ringing, 183, > > answered , cancel and bye etc... > > > > Thank you > > > > Regards, > > Jason > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brett at nemeroff.com Wed Apr 3 21:10:31 2024 From: brett at nemeroff.com (Brett Nemeroff) Date: Wed, 3 Apr 2024 16:10:31 -0500 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: I'd recommend using the events and rabbitmq. You should be able to do just about anything with that. What cps are you processing? On Wed, Apr 3, 2024 at 3:46 PM nz deals wrote: > Thanks Ben, > > The issue with the scheduled task is that it introduces a delay. I'm > exploring methods to enable real-time display. By streaming events directly > from OpenSIPS, we could achieve live updates on the display. > > > Thank you > > Regards, > Jason > > > On Thu, 4 Apr 2024 at 06:15, Ben Newlin wrote: > >> OpenSIPS will already track some very basic statistics like this for you >> using the Statistics module. >> >> >> >> https://opensips.org/docs/modules/3.4.x/statistics.html >> >> >> >> For example, the Dialog module exposes concurrent calls. >> >> >> >> We have a scheduled job that queries those stats via MI and pushes them >> into our external metrics system, allowing us to create dashboards from the >> data. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of nz deals < >> nzdealshelp at gmail.com> >> *Date: *Wednesday, April 3, 2024 at 1:03 PM >> *To: *Bogdan-Andrei Iancu >> *Cc: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] dashboard stats from opensips >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Hi Bogdan, >> >> >> >> Yes, something along the lines of the OpenSIPs control panel, but I'm >> looking for very basic statistics, such as the details of currently active >> calls and a straightforward graph displaying concurrent calls. >> >> >> Thank you >> >> >> >> Regards, >> >> Jason >> >> >> >> On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu >> wrote: >> >> Hi Jason, >> >> Have you checked the Dashboard in OpenSIPS Control Panel ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 03.04.2024 13:54, nz deals wrote: >> > Hi everyone, >> > >> > I'm seeking guidance on creating a dashboard. I'm considering saving >> > dialog events in Redis (straight from OpenSIPS), allowing my dashboard >> > to directly access the Redis cache. Do you think this is a wise >> > strategy, or do you have any alternative suggestions? Any expert's >> > suggestion will be highly appreciated. In fact if someone has any >> > example to check, raise dialog events like call, ringing, 183, >> > answered , cancel and bye etc... >> > >> > Thank you >> > >> > Regards, >> > Jason >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Apr 3 21:37:35 2024 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 4 Apr 2024 10:37:35 +1300 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: Thank you, Brett. My thoughts have been on events and Redis ;) I'll also explore RabbitMQ, thanks for the suggestion. From what I gather, we can utilize E_DLG_STATE_CHANGED within the event route. Could you guide me on how to retrieve the dialogid/callid and its state? If I can access this information in the event_route[E_DLG_STATE_CHANGED], storing the value would be straightforward. Thanks On Thu, 4 Apr 2024 at 10:13, Brett Nemeroff wrote: > I'd recommend using the events and rabbitmq. You should be able to do just > about anything with that. > > What cps are you processing? > > On Wed, Apr 3, 2024 at 3:46 PM nz deals wrote: > >> Thanks Ben, >> >> The issue with the scheduled task is that it introduces a delay. I'm >> exploring methods to enable real-time display. By streaming events directly >> from OpenSIPS, we could achieve live updates on the display. >> >> >> Thank you >> >> Regards, >> Jason >> >> >> On Thu, 4 Apr 2024 at 06:15, Ben Newlin wrote: >> >>> OpenSIPS will already track some very basic statistics like this for you >>> using the Statistics module. >>> >>> >>> >>> https://opensips.org/docs/modules/3.4.x/statistics.html >>> >>> >>> >>> For example, the Dialog module exposes concurrent calls. >>> >>> >>> >>> We have a scheduled job that queries those stats via MI and pushes them >>> into our external metrics system, allowing us to create dashboards from the >>> data. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of nz deals < >>> nzdealshelp at gmail.com> >>> *Date: *Wednesday, April 3, 2024 at 1:03 PM >>> *To: *Bogdan-Andrei Iancu >>> *Cc: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] dashboard stats from opensips >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Hi Bogdan, >>> >>> >>> >>> Yes, something along the lines of the OpenSIPs control panel, but I'm >>> looking for very basic statistics, such as the details of currently active >>> calls and a straightforward graph displaying concurrent calls. >>> >>> >>> Thank you >>> >>> >>> >>> Regards, >>> >>> Jason >>> >>> >>> >>> On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu >>> wrote: >>> >>> Hi Jason, >>> >>> Have you checked the Dashboard in OpenSIPS Control Panel ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 03.04.2024 13:54, nz deals wrote: >>> > Hi everyone, >>> > >>> > I'm seeking guidance on creating a dashboard. I'm considering saving >>> > dialog events in Redis (straight from OpenSIPS), allowing my dashboard >>> > to directly access the Redis cache. Do you think this is a wise >>> > strategy, or do you have any alternative suggestions? Any expert's >>> > suggestion will be highly appreciated. In fact if someone has any >>> > example to check, raise dialog events like call, ringing, 183, >>> > answered , cancel and bye etc... >>> > >>> > Thank you >>> > >>> > Regards, >>> > Jason >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > Users at lists.opensips.org >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 3 22:07:02 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 3 Apr 2024 22:07:02 +0000 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: The parameters exposed by the E_DLG_STATE_CHANGED event are documented [1]. They are accessed using the $param notation [2]. You can then use load_dialog_ctx [3] and get all the other dialog information you need. [1] - https://opensips.org/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED. [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-4#param [3] - https://opensips.org/docs/modules/3.2.x/dialog.html#func_load_dialog_ctx Ben Newlin From: Users on behalf of nz deals Date: Wednesday, April 3, 2024 at 5:38 PM To: OpenSIPS users mailling list , Bogdan-Andrei Iancu Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Thank you, Brett. My thoughts have been on events and Redis ;) I'll also explore RabbitMQ, thanks for the suggestion. From what I gather, we can utilize E_DLG_STATE_CHANGED within the event route. Could you guide me on how to retrieve the dialogid/callid and its state? If I can access this information in the event_route[E_DLG_STATE_CHANGED], storing the value would be straightforward. Thanks On Thu, 4 Apr 2024 at 10:13, Brett Nemeroff > wrote: I'd recommend using the events and rabbitmq. You should be able to do just about anything with that. What cps are you processing? On Wed, Apr 3, 2024 at 3:46 PM nz deals > wrote: Thanks Ben, The issue with the scheduled task is that it introduces a delay. I'm exploring methods to enable real-time display. By streaming events directly from OpenSIPS, we could achieve live updates on the display. Thank you Regards, Jason On Thu, 4 Apr 2024 at 06:15, Ben Newlin > wrote: OpenSIPS will already track some very basic statistics like this for you using the Statistics module. https://opensips.org/docs/modules/3.4.x/statistics.html For example, the Dialog module exposes concurrent calls. We have a scheduled job that queries those stats via MI and pushes them into our external metrics system, allowing us to create dashboards from the data. Ben Newlin From: Users > on behalf of nz deals > Date: Wednesday, April 3, 2024 at 1:03 PM To: Bogdan-Andrei Iancu > Cc: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi Bogdan, Yes, something along the lines of the OpenSIPs control panel, but I'm looking for very basic statistics, such as the details of currently active calls and a straightforward graph displaying concurrent calls. Thank you Regards, Jason On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu > wrote: Hi Jason, Have you checked the Dashboard in OpenSIPS Control Panel ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 13:54, nz deals wrote: > Hi everyone, > > I'm seeking guidance on creating a dashboard. I'm considering saving > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > to directly access the Redis cache. Do you think this is a wise > strategy, or do you have any alternative suggestions? Any expert's > suggestion will be highly appreciated. In fact if someone has any > example to check, raise dialog events like call, ringing, 183, > answered , cancel and bye etc... > > Thank you > > Regards, > Jason > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Apr 3 22:59:26 2024 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 4 Apr 2024 11:59:26 +1300 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: Much appreciated Ben, I was able to test $params so all is good with them. I was not able to access attributes which I have set after the create_dialog. create_dialog(); $dlg_val(caller) = $fu; $dlg_val(callee) = $ru; i also wanted to access $dlg_val(caller) and $dlg_val(callee) under event_route[E_DLG_STATE_CHANGED] On Thu, 4 Apr 2024 at 11:10, Ben Newlin wrote: > The parameters exposed by the E_DLG_STATE_CHANGED event are documented [1]. > > > > They are accessed using the $param notation [2]. > > > > You can then use load_dialog_ctx [3] and get all the other dialog > information you need. > > > > [1] - > https://opensips.org/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED > . > > [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-4#param > > [3] - > https://opensips.org/docs/modules/3.2.x/dialog.html#func_load_dialog_ctx > > > > Ben Newlin > > > > *From: *Users on behalf of nz deals < > nzdealshelp at gmail.com> > *Date: *Wednesday, April 3, 2024 at 5:38 PM > *To: *OpenSIPS users mailling list , > Bogdan-Andrei Iancu > *Subject: *Re: [OpenSIPS-Users] dashboard stats from opensips > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Thank you, Brett. > > My thoughts have been on events and Redis ;) I'll also explore RabbitMQ, > thanks for the suggestion. From what I gather, we can utilize > E_DLG_STATE_CHANGED within the event route. Could you guide me on how to > retrieve the dialogid/callid and its state? If I can access this > information in the event_route[E_DLG_STATE_CHANGED], storing the value > would be straightforward. > > > > Thanks > > > > On Thu, 4 Apr 2024 at 10:13, Brett Nemeroff wrote: > > I'd recommend using the events and rabbitmq. You should be able to do just > about anything with that. > > > > What cps are you processing? > > > > On Wed, Apr 3, 2024 at 3:46 PM nz deals wrote: > > Thanks Ben, > > > > The issue with the scheduled task is that it introduces a delay. I'm > exploring methods to enable real-time display. By streaming events directly > from OpenSIPS, we could achieve live updates on the display. > > > > Thank you > > > > Regards, > > Jason > > > > > > On Thu, 4 Apr 2024 at 06:15, Ben Newlin wrote: > > OpenSIPS will already track some very basic statistics like this for you > using the Statistics module. > > > > https://opensips.org/docs/modules/3.4.x/statistics.html > > > > For example, the Dialog module exposes concurrent calls. > > > > We have a scheduled job that queries those stats via MI and pushes them > into our external metrics system, allowing us to create dashboards from the > data. > > > > Ben Newlin > > > > *From: *Users on behalf of nz deals < > nzdealshelp at gmail.com> > *Date: *Wednesday, April 3, 2024 at 1:03 PM > *To: *Bogdan-Andrei Iancu > *Cc: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] dashboard stats from opensips > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi Bogdan, > > > > Yes, something along the lines of the OpenSIPs control panel, but I'm > looking for very basic statistics, such as the details of currently active > calls and a straightforward graph displaying concurrent calls. > > Thank you > > > > Regards, > > Jason > > > > On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu > wrote: > > Hi Jason, > > Have you checked the Dashboard in OpenSIPS Control Panel ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 03.04.2024 13:54, nz deals wrote: > > Hi everyone, > > > > I'm seeking guidance on creating a dashboard. I'm considering saving > > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > > to directly access the Redis cache. Do you think this is a wise > > strategy, or do you have any alternative suggestions? Any expert's > > suggestion will be highly appreciated. In fact if someone has any > > example to check, raise dialog events like call, ringing, 183, > > answered , cancel and bye etc... > > > > Thank you > > > > Regards, > > Jason > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 4 06:32:31 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 4 Apr 2024 09:32:31 +0300 Subject: [OpenSIPS-Users] memory fragmentation when calling dr_reload. In-Reply-To: References: Message-ID: <6bf23123-7cc4-4eb9-9951-e7b8ef798881@opensips.org> That's EOL for quite some time :( Either consider more pkg mem (to cope with fragmentation) , either an upgrade to 3.4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 17:34, Johan De Clercq wrote: > A very old one 2.2.7 > > Op wo 3 apr 2024 om 16:26 schreef Bogdan-Andrei Iancu > : > > What OpenSIPS version is there ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 03.04.2024 17:04, Johan De Clercq wrote: >> In addtion, I have 24 children, so can I increase in some way >> only the process that handles the fifo requests ? >> >> Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq : >> >> Hi, >> >> A client has a very big dynamic routing rule set. (dr_rules >> >= 2.1 gb ). >> When reloading the db in opensips (dr_reload), I see below >> error in the log >> >> ERROR:core:fm_malloc: not enough free pkg memory (268008864 >> bytes left), please increase the "-M" command line parameter! >> the -M parameter is now at 256. Should I increase this to >> 3000 to work aroun this issue ? >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 4 06:32:33 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 4 Apr 2024 12:02:33 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B wrote: > Yes. I am canceling the call prior to dialog creation. > > Sent from Outlook for Android > ------------------------------ > *From:* Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Sent:* Wednesday, April 3, 2024 7:03:33 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > > If your script is cancelling the call then why wouldn’t you “capture it” > in the same place? Send whatever you need to whatever external entity you > are using directly. You don’t need a callback to trigger if you know you > are taking the action. > > > > A created dialog being cancelled should result in a state change event – > to CANCELLED I think - so I assume you mean you are cancelling it prior to > dialog creation, in which case there won’t be any dialog callback. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Wednesday, April 3, 2024 at 2:35 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I am capturing the dropped call after ringing in failure_route, but If I > cancelled during the start of the call, how to capture it. > > > > On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: > > The start of the call would be when you call “create_dialog”. The dialog > state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change > event is raised for creation. It may only be raised when the state changes > after creation. But since you control the dialog creation, you can just > take whatever action you desire at that time. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Tuesday, April 2, 2024 at 8:05 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I tried is_method("INVITE"), but it is getting called only at the start of > RINGING. > > > > On Tue, 2 Apr 2024 at 15:09, Prathibha B wrote: > > I am able to capture the trying status also. But not getting the START of > the call... > > > > On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > > How do I identify the START and TRYING state of the call? > > > > I am able to capture RINGING, ANSWER and TERMINATED states. > > > > On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > > I tried > > event_route[E_DLG_STATE_CHANGED] { > > } > > > > I am getting syntax error. > > > > On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > > How to use *E_DLG_STATE_CHANGED to identify the start of the call?* > > > > > > On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > > You can also use the REST client. And there are many other ways, as well. > > > > There is no single correct answer to the vague question of connecting to > any generic “external application”. You must understand your systems and > decide the best approach depending on the needs and capabilities of both > the external application and OpenSIPS. > > > > Ben Newlin > > > > *From: *Users on behalf of > Bogdan-Andrei Iancu > *Date: *Wednesday, March 20, 2024 at 10:06 AM > *To: *OpenSIPS users mailling list , Prathibha > B > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Use the dialog events: > > https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED > > And you subscribe from outside OpenSIPS for such events: > https://www.opensips.org/Documentation/Interface-Events-3-4 > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 20.03.2024 12:16, Prathibha B wrote: > > No. I want to pass START, CONNECT, END messages from OpenSIPS to external > application. > > > > On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: > > Well, to execute external command from opensips you may want to use EXEC > module. > > this is a manual for v3.2: > > https://opensips.org/html/docs/modules/3.2.x/exec.html > > > > On 3/20/2024 5:00 AM, Prathibha B wrote: > > How to integrate OpenSIPS with external applications? > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Best Regards: > > Marcin Groszek > > Business Phone Service > > https://www.voipplus.net > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 4 06:33:25 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 4 Apr 2024 12:03:25 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > if($rm == "INVITE") { > xlog("Request method = $rm"); > exec("script.sh \"INVITE\""); > } > > > With the above code , I am getting the Request method = INVITE twice in > the log file. But the exec() is not getting executed. > > > > On Wed, 3 Apr 2024 at 19:43, Prathibha B wrote: > >> Yes. I am canceling the call prior to dialog creation. >> >> Sent from Outlook for Android >> ------------------------------ >> *From:* Users on behalf of Ben Newlin >> >> *Sent:* Wednesday, April 3, 2024 7:03:33 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] external applications >> >> >> If your script is cancelling the call then why wouldn’t you “capture it” >> in the same place? Send whatever you need to whatever external entity you >> are using directly. You don’t need a callback to trigger if you know you >> are taking the action. >> >> >> >> A created dialog being cancelled should result in a state change event – >> to CANCELLED I think - so I assume you mean you are cancelling it prior to >> dialog creation, in which case there won’t be any dialog callback. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Wednesday, April 3, 2024 at 2:35 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I am capturing the dropped call after ringing in failure_route, but If I >> cancelled during the start of the call, how to capture it. >> >> >> >> On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: >> >> The start of the call would be when you call “create_dialog”. The dialog >> state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change >> event is raised for creation. It may only be raised when the state changes >> after creation. But since you control the dialog creation, you can just >> take whatever action you desire at that time. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Tuesday, April 2, 2024 at 8:05 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I tried is_method("INVITE"), but it is getting called only at the start >> of RINGING. >> >> >> >> On Tue, 2 Apr 2024 at 15:09, Prathibha B >> wrote: >> >> I am able to capture the trying status also. But not getting the START of >> the call... >> >> >> >> On Tue, 2 Apr 2024 at 14:59, Prathibha B >> wrote: >> >> How do I identify the START and TRYING state of the call? >> >> >> >> I am able to capture RINGING, ANSWER and TERMINATED states. >> >> >> >> On Tue, 2 Apr 2024 at 14:51, Prathibha B >> wrote: >> >> I tried >> >> event_route[E_DLG_STATE_CHANGED] { >> >> } >> >> >> >> I am getting syntax error. >> >> >> >> On Tue, 2 Apr 2024 at 14:45, Prathibha B >> wrote: >> >> How to use *E_DLG_STATE_CHANGED to identify the start of the call?* >> >> >> >> >> >> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >> >> You can also use the REST client. And there are many other ways, as well. >> >> >> >> There is no single correct answer to the vague question of connecting to >> any generic “external application”. You must understand your systems and >> decide the best approach depending on the needs and capabilities of both >> the external application and OpenSIPS. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of >> Bogdan-Andrei Iancu >> *Date: *Wednesday, March 20, 2024 at 10:06 AM >> *To: *OpenSIPS users mailling list , Prathibha >> B >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Use the dialog events: >> >> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >> >> And you subscribe from outside OpenSIPS for such events: >> https://www.opensips.org/Documentation/Interface-Events-3-4 >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> https://www.opensips-solutions.com >> >> https://www.siphub.com >> >> On 20.03.2024 12:16, Prathibha B wrote: >> >> No. I want to pass START, CONNECT, END messages from OpenSIPS to external >> application. >> >> >> >> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: >> >> Well, to execute external command from opensips you may want to use EXEC >> module. >> >> this is a manual for v3.2: >> >> https://opensips.org/html/docs/modules/3.2.x/exec.html >> >> >> >> On 3/20/2024 5:00 AM, Prathibha B wrote: >> >> How to integrate OpenSIPS with external applications? >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> >> Best Regards: >> >> Marcin Groszek >> >> Business Phone Service >> >> https://www.voipplus.net >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 4 06:43:02 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 4 Apr 2024 12:13:02 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I am not getting the log also. On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > if($rm == "INVITE") { > xlog("Request method = $rm"); > exec("script.sh \"INVITE\""); > } > > > With the above code , I am getting the Request method = INVITE twice in > the log file. But the exec() is not getting executed. > > > > On Wed, 3 Apr 2024 at 19:43, Prathibha B wrote: > >> Yes. I am canceling the call prior to dialog creation. >> >> Sent from Outlook for Android >> ------------------------------ >> *From:* Users on behalf of Ben Newlin >> >> *Sent:* Wednesday, April 3, 2024 7:03:33 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] external applications >> >> >> If your script is cancelling the call then why wouldn’t you “capture it” >> in the same place? Send whatever you need to whatever external entity you >> are using directly. You don’t need a callback to trigger if you know you >> are taking the action. >> >> >> >> A created dialog being cancelled should result in a state change event – >> to CANCELLED I think - so I assume you mean you are cancelling it prior to >> dialog creation, in which case there won’t be any dialog callback. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Wednesday, April 3, 2024 at 2:35 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I am capturing the dropped call after ringing in failure_route, but If I >> cancelled during the start of the call, how to capture it. >> >> >> >> On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: >> >> The start of the call would be when you call “create_dialog”. The dialog >> state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change >> event is raised for creation. It may only be raised when the state changes >> after creation. But since you control the dialog creation, you can just >> take whatever action you desire at that time. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Tuesday, April 2, 2024 at 8:05 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I tried is_method("INVITE"), but it is getting called only at the start >> of RINGING. >> >> >> >> On Tue, 2 Apr 2024 at 15:09, Prathibha B >> wrote: >> >> I am able to capture the trying status also. But not getting the START of >> the call... >> >> >> >> On Tue, 2 Apr 2024 at 14:59, Prathibha B >> wrote: >> >> How do I identify the START and TRYING state of the call? >> >> >> >> I am able to capture RINGING, ANSWER and TERMINATED states. >> >> >> >> On Tue, 2 Apr 2024 at 14:51, Prathibha B >> wrote: >> >> I tried >> >> event_route[E_DLG_STATE_CHANGED] { >> >> } >> >> >> >> I am getting syntax error. >> >> >> >> On Tue, 2 Apr 2024 at 14:45, Prathibha B >> wrote: >> >> How to use *E_DLG_STATE_CHANGED to identify the start of the call?* >> >> >> >> >> >> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >> >> You can also use the REST client. And there are many other ways, as well. >> >> >> >> There is no single correct answer to the vague question of connecting to >> any generic “external application”. You must understand your systems and >> decide the best approach depending on the needs and capabilities of both >> the external application and OpenSIPS. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of >> Bogdan-Andrei Iancu >> *Date: *Wednesday, March 20, 2024 at 10:06 AM >> *To: *OpenSIPS users mailling list , Prathibha >> B >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Use the dialog events: >> >> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >> >> And you subscribe from outside OpenSIPS for such events: >> https://www.opensips.org/Documentation/Interface-Events-3-4 >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> https://www.opensips-solutions.com >> >> https://www.siphub.com >> >> On 20.03.2024 12:16, Prathibha B wrote: >> >> No. I want to pass START, CONNECT, END messages from OpenSIPS to external >> application. >> >> >> >> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: >> >> Well, to execute external command from opensips you may want to use EXEC >> module. >> >> this is a manual for v3.2: >> >> https://opensips.org/html/docs/modules/3.2.x/exec.html >> >> >> >> On 3/20/2024 5:00 AM, Prathibha B wrote: >> >> How to integrate OpenSIPS with external applications? >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> >> Best Regards: >> >> Marcin Groszek >> >> Business Phone Service >> >> https://www.voipplus.net >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 4 06:48:36 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 4 Apr 2024 12:18:36 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: However this code is getting executed after reaching the ringing state. On Thu, 4 Apr 2024 at 12:13, Prathibha B wrote: > I am not getting the log also. > > On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > >> if($rm == "INVITE") { >> xlog("Request method = $rm"); >> exec("script.sh \"INVITE\""); >> } >> >> >> With the above code , I am getting the Request method = INVITE twice in >> the log file. But the exec() is not getting executed. >> >> >> >> On Wed, 3 Apr 2024 at 19:43, Prathibha B >> wrote: >> >>> Yes. I am canceling the call prior to dialog creation. >>> >>> Sent from Outlook for Android >>> ------------------------------ >>> *From:* Users on behalf of Ben >>> Newlin >>> *Sent:* Wednesday, April 3, 2024 7:03:33 PM >>> *To:* OpenSIPS users mailling list >>> *Subject:* Re: [OpenSIPS-Users] external applications >>> >>> >>> If your script is cancelling the call then why wouldn’t you “capture it” >>> in the same place? Send whatever you need to whatever external entity you >>> are using directly. You don’t need a callback to trigger if you know you >>> are taking the action. >>> >>> >>> >>> A created dialog being cancelled should result in a state change event – >>> to CANCELLED I think - so I assume you mean you are cancelling it prior to >>> dialog creation, in which case there won’t be any dialog callback. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Prathibha >>> B >>> *Date: *Wednesday, April 3, 2024 at 2:35 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] external applications >>> >>> >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> I am capturing the dropped call after ringing in failure_route, but If >>> I cancelled during the start of the call, how to capture it. >>> >>> >>> >>> On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: >>> >>> The start of the call would be when you call “create_dialog”. The dialog >>> state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change >>> event is raised for creation. It may only be raised when the state changes >>> after creation. But since you control the dialog creation, you can just >>> take whatever action you desire at that time. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Prathibha >>> B >>> *Date: *Tuesday, April 2, 2024 at 8:05 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] external applications >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> I tried is_method("INVITE"), but it is getting called only at the start >>> of RINGING. >>> >>> >>> >>> On Tue, 2 Apr 2024 at 15:09, Prathibha B >>> wrote: >>> >>> I am able to capture the trying status also. But not getting the START >>> of the call... >>> >>> >>> >>> On Tue, 2 Apr 2024 at 14:59, Prathibha B >>> wrote: >>> >>> How do I identify the START and TRYING state of the call? >>> >>> >>> >>> I am able to capture RINGING, ANSWER and TERMINATED states. >>> >>> >>> >>> On Tue, 2 Apr 2024 at 14:51, Prathibha B >>> wrote: >>> >>> I tried >>> >>> event_route[E_DLG_STATE_CHANGED] { >>> >>> } >>> >>> >>> >>> I am getting syntax error. >>> >>> >>> >>> On Tue, 2 Apr 2024 at 14:45, Prathibha B >>> wrote: >>> >>> How to use *E_DLG_STATE_CHANGED to identify the start of the call?* >>> >>> >>> >>> >>> >>> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >>> >>> You can also use the REST client. And there are many other ways, as well. >>> >>> >>> >>> There is no single correct answer to the vague question of connecting to >>> any generic “external application”. You must understand your systems and >>> decide the best approach depending on the needs and capabilities of both >>> the external application and OpenSIPS. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of >>> Bogdan-Andrei Iancu >>> *Date: *Wednesday, March 20, 2024 at 10:06 AM >>> *To: *OpenSIPS users mailling list , >>> Prathibha B >>> *Subject: *Re: [OpenSIPS-Users] external applications >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Use the dialog events: >>> >>> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >>> >>> And you subscribe from outside OpenSIPS for such events: >>> https://www.opensips.org/Documentation/Interface-Events-3-4 >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> >>> >>> OpenSIPS Founder and Developer >>> >>> https://www.opensips-solutions.com >>> >>> https://www.siphub.com >>> >>> On 20.03.2024 12:16, Prathibha B wrote: >>> >>> No. I want to pass START, CONNECT, END messages from OpenSIPS to >>> external application. >>> >>> >>> >>> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek >>> wrote: >>> >>> Well, to execute external command from opensips you may want to use EXEC >>> module. >>> >>> this is a manual for v3.2: >>> >>> https://opensips.org/html/docs/modules/3.2.x/exec.html >>> >>> >>> >>> On 3/20/2024 5:00 AM, Prathibha B wrote: >>> >>> How to integrate OpenSIPS with external applications? >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> _______________________________________________ >>> >>> Users mailing list >>> >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> >>> Best Regards: >>> >>> Marcin Groszek >>> >>> Business Phone Service >>> >>> https://www.voipplus.net >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> _______________________________________________ >>> >>> Users mailing list >>> >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Apr 4 10:45:08 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 4 Apr 2024 16:15:08 +0530 Subject: [OpenSIPS-Users] socket defination in opensips 3.2 Message-ID: Hi All , Is the below socket definition right ? In this case opensips will listen on both UDp and TCP protocol on the same 5507 port ? socket=tcp:10.5.1.1:5507 socket=udp:10.5.1.1:5507 *If this is not right then how can opensips listen for both UDP and TCP protocol in a single port ?* *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Apr 4 10:51:10 2024 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 4 Apr 2024 06:51:10 -0400 Subject: [OpenSIPS-Users] socket defination in opensips 3.2 In-Reply-To: References: Message-ID: Hello, Port spaces for UDP and TCP are independent. -- Alex > On Apr 4, 2024, at 6:45 AM, Sasmita Panda wrote: > > Hi All , > > Is the below socket definition right ? In this case opensips will listen on both UDp and TCP protocol on the same 5507 port ? > > socket=tcp:10.5.1.1:5507 > socket=udp:10.5.1.1:5507 > > If this is not right then how can opensips listen for both UDP and TCP protocol in a single port ? > > Thanks & Regards > Sasmita Panda > Senior Network Testing and Software Engineer > 3CLogic , ph:07827611765 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 From prathibhab.tvm at gmail.com Thu Apr 4 11:28:34 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 4 Apr 2024 16:58:34 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: what does the $rm = SUBSCRIBE mean? On Thu, 4 Apr 2024 at 12:18, Prathibha B wrote: > However this code is getting executed after reaching the ringing state. > > On Thu, 4 Apr 2024 at 12:13, Prathibha B wrote: > >> I am not getting the log also. >> >> On Thu, 4 Apr 2024 at 12:02, Prathibha B >> wrote: >> >>> if($rm == "INVITE") { >>> xlog("Request method = $rm"); >>> exec("script.sh \"INVITE\""); >>> } >>> >>> >>> With the above code , I am getting the Request method = INVITE twice in >>> the log file. But the exec() is not getting executed. >>> >>> >>> >>> On Wed, 3 Apr 2024 at 19:43, Prathibha B >>> wrote: >>> >>>> Yes. I am canceling the call prior to dialog creation. >>>> >>>> Sent from Outlook for Android >>>> ------------------------------ >>>> *From:* Users on behalf of Ben >>>> Newlin >>>> *Sent:* Wednesday, April 3, 2024 7:03:33 PM >>>> *To:* OpenSIPS users mailling list >>>> *Subject:* Re: [OpenSIPS-Users] external applications >>>> >>>> >>>> If your script is cancelling the call then why wouldn’t you “capture >>>> it” in the same place? Send whatever you need to whatever external entity >>>> you are using directly. You don’t need a callback to trigger if you know >>>> you are taking the action. >>>> >>>> >>>> >>>> A created dialog being cancelled should result in a state change event >>>> – to CANCELLED I think - so I assume you mean you are cancelling it prior >>>> to dialog creation, in which case there won’t be any dialog callback. >>>> >>>> >>>> >>>> Ben Newlin >>>> >>>> >>>> >>>> *From: *Users on behalf of >>>> Prathibha B >>>> *Date: *Wednesday, April 3, 2024 at 2:35 AM >>>> *To: *OpenSIPS users mailling list >>>> *Subject: *Re: [OpenSIPS-Users] external applications >>>> >>>> >>>> >>>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>>> >>>> >>>> ------------------------------ >>>> >>>> I am capturing the dropped call after ringing in failure_route, but If >>>> I cancelled during the start of the call, how to capture it. >>>> >>>> >>>> >>>> On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: >>>> >>>> The start of the call would be when you call “create_dialog”. The >>>> dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state >>>> change event is raised for creation. It may only be raised when the state >>>> changes after creation. But since you control the dialog creation, you can >>>> just take whatever action you desire at that time. >>>> >>>> >>>> >>>> Ben Newlin >>>> >>>> >>>> >>>> *From: *Users on behalf of >>>> Prathibha B >>>> *Date: *Tuesday, April 2, 2024 at 8:05 AM >>>> *To: *OpenSIPS users mailling list >>>> *Subject: *Re: [OpenSIPS-Users] external applications >>>> >>>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>>> >>>> >>>> ------------------------------ >>>> >>>> I tried is_method("INVITE"), but it is getting called only at the start >>>> of RINGING. >>>> >>>> >>>> >>>> On Tue, 2 Apr 2024 at 15:09, Prathibha B >>>> wrote: >>>> >>>> I am able to capture the trying status also. But not getting the START >>>> of the call... >>>> >>>> >>>> >>>> On Tue, 2 Apr 2024 at 14:59, Prathibha B >>>> wrote: >>>> >>>> How do I identify the START and TRYING state of the call? >>>> >>>> >>>> >>>> I am able to capture RINGING, ANSWER and TERMINATED states. >>>> >>>> >>>> >>>> On Tue, 2 Apr 2024 at 14:51, Prathibha B >>>> wrote: >>>> >>>> I tried >>>> >>>> event_route[E_DLG_STATE_CHANGED] { >>>> >>>> } >>>> >>>> >>>> >>>> I am getting syntax error. >>>> >>>> >>>> >>>> On Tue, 2 Apr 2024 at 14:45, Prathibha B >>>> wrote: >>>> >>>> How to use *E_DLG_STATE_CHANGED to identify the start of the call?* >>>> >>>> >>>> >>>> >>>> >>>> On Wed, 20 Mar 2024 at 19:46, Ben Newlin >>>> wrote: >>>> >>>> You can also use the REST client. And there are many other ways, as >>>> well. >>>> >>>> >>>> >>>> There is no single correct answer to the vague question of connecting >>>> to any generic “external application”. You must understand your systems and >>>> decide the best approach depending on the needs and capabilities of both >>>> the external application and OpenSIPS. >>>> >>>> >>>> >>>> Ben Newlin >>>> >>>> >>>> >>>> *From: *Users on behalf of >>>> Bogdan-Andrei Iancu >>>> *Date: *Wednesday, March 20, 2024 at 10:06 AM >>>> *To: *OpenSIPS users mailling list , >>>> Prathibha B >>>> *Subject: *Re: [OpenSIPS-Users] external applications >>>> >>>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>>> >>>> >>>> ------------------------------ >>>> >>>> Use the dialog events: >>>> >>>> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >>>> >>>> And you subscribe from outside OpenSIPS for such events: >>>> https://www.opensips.org/Documentation/Interface-Events-3-4 >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> >>>> >>>> OpenSIPS Founder and Developer >>>> >>>> https://www.opensips-solutions.com >>>> >>>> https://www.siphub.com >>>> >>>> On 20.03.2024 12:16, Prathibha B wrote: >>>> >>>> No. I want to pass START, CONNECT, END messages from OpenSIPS to >>>> external application. >>>> >>>> >>>> >>>> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek >>>> wrote: >>>> >>>> Well, to execute external command from opensips you may want to use >>>> EXEC module. >>>> >>>> this is a manual for v3.2: >>>> >>>> https://opensips.org/html/docs/modules/3.2.x/exec.html >>>> >>>> >>>> >>>> On 3/20/2024 5:00 AM, Prathibha B wrote: >>>> >>>> How to integrate OpenSIPS with external applications? >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> Users mailing list >>>> >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> -- >>>> >>>> Best Regards: >>>> >>>> Marcin Groszek >>>> >>>> Business Phone Service >>>> >>>> https://www.voipplus.net >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> Users mailing list >>>> >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 4 11:50:22 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 4 Apr 2024 14:50:22 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024 - Speaker's lineup Message-ID: OpenSIPS Summit May 14th - 17th, 2024 Valencia, Spain *Speaker's lineup & Schedule * We bring here the list of speakers and papers - great speakers presenting great topics to share experience and knowledge to a great audience. Explore here all the details... *Attend to learn* - the registration process is ongoing, the training class is almost full, so hurry up. The/*Corporate Package*/ is available with an attractive discount. Register now ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Apr 4 15:52:28 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 4 Apr 2024 15:52:28 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: Per the documentation [1], input variables to the script are provided as a separate parameter to the exec command. It does not do a direct shell execution of the entire first parameter. So it should be: exec(“script.sh”, “INVITE”); Also, for a relative command I’m not entirely sure where OpenSIPS would look. It could be the working directory or it could be the directory from which opensips was launched. Have you tried using an absolute path? Are you setting the wdir [2] parameter? [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec [2] - https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B > wrote: if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B > wrote: Yes. I am canceling the call prior to dialog creation. Sent from Outlook for Android ________________________________ From: Users > on behalf of Ben Newlin > Sent: Wednesday, April 3, 2024 7:03:33 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users > on behalf of Prathibha B > Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 5 00:19:37 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 5 Apr 2024 00:19:37 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: My question is not regarding exec command. How to capture the start of the call? I'm unable to get the start. Sent from Outlook for Android ________________________________ From: Users on behalf of Ben Newlin Sent: Thursday, April 4, 2024 9:22:28 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications Per the documentation [1], input variables to the script are provided as a separate parameter to the exec command. It does not do a direct shell execution of the entire first parameter. So it should be: exec(“script.sh”, “INVITE”); Also, for a relative command I’m not entirely sure where OpenSIPS would look. It could be the working directory or it could be the directory from which opensips was launched. Have you tried using an absolute path? Are you setting the wdir [2] parameter? [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec [2] - https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B > wrote: if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B > wrote: Yes. I am canceling the call prior to dialog creation. Sent from Outlook for Android ________________________________ From: Users > on behalf of Ben Newlin > Sent: Wednesday, April 3, 2024 7:03:33 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users > on behalf of Prathibha B > Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 5 00:21:05 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 5 Apr 2024 00:21:05 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: Opensips exec() is working using the absolute path. Sent from Outlook for Android ________________________________ From: Prathibha B Sent: Friday, April 5, 2024 5:49:37 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications My question is not regarding exec command. How to capture the start of the call? I'm unable to get the start. Sent from Outlook for Android ________________________________ From: Users on behalf of Ben Newlin Sent: Thursday, April 4, 2024 9:22:28 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications Per the documentation [1], input variables to the script are provided as a separate parameter to the exec command. It does not do a direct shell execution of the entire first parameter. So it should be: exec(“script.sh”, “INVITE”); Also, for a relative command I’m not entirely sure where OpenSIPS would look. It could be the working directory or it could be the directory from which opensips was launched. Have you tried using an absolute path? Are you setting the wdir [2] parameter? [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec [2] - https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B > wrote: if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B > wrote: Yes. I am canceling the call prior to dialog creation. Sent from Outlook for Android ________________________________ From: Users > on behalf of Ben Newlin > Sent: Wednesday, April 3, 2024 7:03:33 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users > on behalf of Prathibha B > Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Apr 5 06:01:03 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 5 Apr 2024 11:31:03 +0530 Subject: [OpenSIPS-Users] socket defination in opensips 3.2 In-Reply-To: References: Message-ID: Hi Alex , Thank you for your reply . I have tested this and it's working for me as well . Before moving to this solution earlier I was using TCP proxy but I am facing one issue . *UAC1-Invite (CSeq - 10428)-TCP proxy-UDP - TCP -- INVITE (**CSeq - 10428**)- UAC2* *200 Ok * *UAC1 - ACK(CSeq-10428)- TCP proxy-UDP-TCP Proxy - (ACK with CSeq 10429)UAC2 * *UAC1- Re-Invite (CSeq-10429) - TCP proxy - UDP - TCP proxy - UAC2 * *Here the ACK when reaches UAC2 the Cseq gets incremented by 1 and also the Re-Invite has the same CSeq . So the UAC2 is rejected with "500 Invalid CSeq" and the call gets disconnected here . * *In the live environment out of 6000 calls I am getting 2/3 calls everyday which are failing for this reason . How do I resolve this ?* *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Apr 4, 2024 at 4:23 PM Alex Balashov wrote: > Hello, > > Port spaces for UDP and TCP are independent. > > -- Alex > > > On Apr 4, 2024, at 6:45 AM, Sasmita Panda wrote: > > > > Hi All , > > > > Is the below socket definition right ? In this case opensips will listen > on both UDp and TCP protocol on the same 5507 port ? > > > > socket=tcp:10.5.1.1:5507 > > socket=udp:10.5.1.1:5507 > > > > If this is not right then how can opensips listen for both UDP and TCP > protocol in a single port ? > > > > Thanks & Regards > > Sasmita Panda > > Senior Network Testing and Software Engineer > > 3CLogic , ph:07827611765 > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 5 07:39:03 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 5 Apr 2024 13:09:03 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: I am not getting $DLG_status = UNCONFORMED at the start of the call. Can someone help? On Fri, 5 Apr 2024 at 05:51, Prathibha B wrote: > Opensips exec() is working using the absolute path. > > Sent from Outlook for Android > ------------------------------ > *From:* Prathibha B > *Sent:* Friday, April 5, 2024 5:49:37 AM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > My question is not regarding exec command. How to capture the start of the > call? I'm unable to get the start. > > Sent from Outlook for Android > ------------------------------ > *From:* Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Sent:* Thursday, April 4, 2024 9:22:28 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > > Per the documentation [1], input variables to the script are provided as a > separate parameter to the exec command. It does not do a direct shell > execution of the entire first parameter. So it should be: > > > > exec(“script.sh”, “INVITE”); > > > > Also, for a relative command I’m not entirely sure where OpenSIPS would > look. It could be the working directory or it could be the directory from > which opensips was launched. Have you tried using an absolute path? Are you > setting the wdir [2] parameter? > > > > [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec > > [2] - > https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Thursday, April 4, 2024 at 11:32 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I've used the above code inside the route block. > > > > On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > > if($rm == "INVITE") { > > xlog("Request method = $rm"); > > exec("script.sh \"INVITE\""); > > } > > > > > > With the above code , I am getting the Request method = INVITE twice in > the log file. But the exec() is not getting executed. > > > > > > > > On Wed, 3 Apr 2024 at 19:43, Prathibha B wrote: > > Yes. I am canceling the call prior to dialog creation. > > > > Sent from Outlook for Android > ------------------------------ > > *From:* Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Sent:* Wednesday, April 3, 2024 7:03:33 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > > > If your script is cancelling the call then why wouldn’t you “capture it” > in the same place? Send whatever you need to whatever external entity you > are using directly. You don’t need a callback to trigger if you know you > are taking the action. > > > > A created dialog being cancelled should result in a state change event – > to CANCELLED I think - so I assume you mean you are cancelling it prior to > dialog creation, in which case there won’t be any dialog callback. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Wednesday, April 3, 2024 at 2:35 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I am capturing the dropped call after ringing in failure_route, but If I > cancelled during the start of the call, how to capture it. > > > > On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: > > The start of the call would be when you call “create_dialog”. The dialog > state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change > event is raised for creation. It may only be raised when the state changes > after creation. But since you control the dialog creation, you can just > take whatever action you desire at that time. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Tuesday, April 2, 2024 at 8:05 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > I tried is_method("INVITE"), but it is getting called only at the start of > RINGING. > > > > On Tue, 2 Apr 2024 at 15:09, Prathibha B wrote: > > I am able to capture the trying status also. But not getting the START of > the call... > > > > On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > > How do I identify the START and TRYING state of the call? > > > > I am able to capture RINGING, ANSWER and TERMINATED states. > > > > On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > > I tried > > event_route[E_DLG_STATE_CHANGED] { > > } > > > > I am getting syntax error. > > > > On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > > How to use *E_DLG_STATE_CHANGED to identify the start of the call?* > > > > > > On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > > You can also use the REST client. And there are many other ways, as well. > > > > There is no single correct answer to the vague question of connecting to > any generic “external application”. You must understand your systems and > decide the best approach depending on the needs and capabilities of both > the external application and OpenSIPS. > > > > Ben Newlin > > > > *From: *Users on behalf of > Bogdan-Andrei Iancu > *Date: *Wednesday, March 20, 2024 at 10:06 AM > *To: *OpenSIPS users mailling list , Prathibha > B > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Use the dialog events: > > https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED > > And you subscribe from outside OpenSIPS for such events: > https://www.opensips.org/Documentation/Interface-Events-3-4 > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 20.03.2024 12:16, Prathibha B wrote: > > No. I want to pass START, CONNECT, END messages from OpenSIPS to external > application. > > > > On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: > > Well, to execute external command from opensips you may want to use EXEC > module. > > this is a manual for v3.2: > > https://opensips.org/html/docs/modules/3.2.x/exec.html > > > > On 3/20/2024 5:00 AM, Prathibha B wrote: > > How to integrate OpenSIPS with external applications? > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Best Regards: > > Marcin Groszek > > Business Phone Service > > https://www.voipplus.net > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 5 07:41:04 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 5 Apr 2024 13:11:04 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: route { if($DLG_status == 1) xlog("UNCONFIRMED\n"); } On Fri, 5 Apr 2024 at 13:09, Prathibha B wrote: > I am not getting $DLG_status = UNCONFORMED at the start of the call. Can > someone help? > > On Fri, 5 Apr 2024 at 05:51, Prathibha B wrote: > >> Opensips exec() is working using the absolute path. >> >> Sent from Outlook for Android >> ------------------------------ >> *From:* Prathibha B >> *Sent:* Friday, April 5, 2024 5:49:37 AM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] external applications >> >> My question is not regarding exec command. How to capture the start of >> the call? I'm unable to get the start. >> >> Sent from Outlook for Android >> ------------------------------ >> *From:* Users on behalf of Ben Newlin >> >> *Sent:* Thursday, April 4, 2024 9:22:28 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] external applications >> >> >> Per the documentation [1], input variables to the script are provided as >> a separate parameter to the exec command. It does not do a direct shell >> execution of the entire first parameter. So it should be: >> >> >> >> exec(“script.sh”, “INVITE”); >> >> >> >> Also, for a relative command I’m not entirely sure where OpenSIPS would >> look. It could be the working directory or it could be the directory from >> which opensips was launched. Have you tried using an absolute path? Are you >> setting the wdir [2] parameter? >> >> >> >> [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec >> >> [2] - >> https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Thursday, April 4, 2024 at 11:32 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I've used the above code inside the route block. >> >> >> >> On Thu, 4 Apr 2024 at 12:02, Prathibha B >> wrote: >> >> if($rm == "INVITE") { >> >> xlog("Request method = $rm"); >> >> exec("script.sh \"INVITE\""); >> >> } >> >> >> >> >> >> With the above code , I am getting the Request method = INVITE twice in >> the log file. But the exec() is not getting executed. >> >> >> >> >> >> >> >> On Wed, 3 Apr 2024 at 19:43, Prathibha B >> wrote: >> >> Yes. I am canceling the call prior to dialog creation. >> >> >> >> Sent from Outlook for Android >> ------------------------------ >> >> *From:* Users on behalf of Ben Newlin >> >> *Sent:* Wednesday, April 3, 2024 7:03:33 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] external applications >> >> >> >> If your script is cancelling the call then why wouldn’t you “capture it” >> in the same place? Send whatever you need to whatever external entity you >> are using directly. You don’t need a callback to trigger if you know you >> are taking the action. >> >> >> >> A created dialog being cancelled should result in a state change event – >> to CANCELLED I think - so I assume you mean you are cancelling it prior to >> dialog creation, in which case there won’t be any dialog callback. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Wednesday, April 3, 2024 at 2:35 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I am capturing the dropped call after ringing in failure_route, but If I >> cancelled during the start of the call, how to capture it. >> >> >> >> On Tue, 2 Apr 2024 at 20:38, Ben Newlin wrote: >> >> The start of the call would be when you call “create_dialog”. The dialog >> state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change >> event is raised for creation. It may only be raised when the state changes >> after creation. But since you control the dialog creation, you can just >> take whatever action you desire at that time. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Prathibha >> B >> *Date: *Tuesday, April 2, 2024 at 8:05 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> I tried is_method("INVITE"), but it is getting called only at the start >> of RINGING. >> >> >> >> On Tue, 2 Apr 2024 at 15:09, Prathibha B >> wrote: >> >> I am able to capture the trying status also. But not getting the START of >> the call... >> >> >> >> On Tue, 2 Apr 2024 at 14:59, Prathibha B >> wrote: >> >> How do I identify the START and TRYING state of the call? >> >> >> >> I am able to capture RINGING, ANSWER and TERMINATED states. >> >> >> >> On Tue, 2 Apr 2024 at 14:51, Prathibha B >> wrote: >> >> I tried >> >> event_route[E_DLG_STATE_CHANGED] { >> >> } >> >> >> >> I am getting syntax error. >> >> >> >> On Tue, 2 Apr 2024 at 14:45, Prathibha B >> wrote: >> >> How to use *E_DLG_STATE_CHANGED to identify the start of the call?* >> >> >> >> >> >> On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: >> >> You can also use the REST client. And there are many other ways, as well. >> >> >> >> There is no single correct answer to the vague question of connecting to >> any generic “external application”. You must understand your systems and >> decide the best approach depending on the needs and capabilities of both >> the external application and OpenSIPS. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of >> Bogdan-Andrei Iancu >> *Date: *Wednesday, March 20, 2024 at 10:06 AM >> *To: *OpenSIPS users mailling list , Prathibha >> B >> *Subject: *Re: [OpenSIPS-Users] external applications >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Use the dialog events: >> >> https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED >> >> And you subscribe from outside OpenSIPS for such events: >> https://www.opensips.org/Documentation/Interface-Events-3-4 >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> https://www.opensips-solutions.com >> >> https://www.siphub.com >> >> On 20.03.2024 12:16, Prathibha B wrote: >> >> No. I want to pass START, CONNECT, END messages from OpenSIPS to external >> application. >> >> >> >> On Wed, 20 Mar 2024 at 15:42, Marcin Groszek wrote: >> >> Well, to execute external command from opensips you may want to use EXEC >> module. >> >> this is a manual for v3.2: >> >> https://opensips.org/html/docs/modules/3.2.x/exec.html >> >> >> >> On 3/20/2024 5:00 AM, Prathibha B wrote: >> >> How to integrate OpenSIPS with external applications? >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> >> Best Regards: >> >> Marcin Groszek >> >> Business Phone Service >> >> https://www.voipplus.net >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 5 08:53:33 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 5 Apr 2024 11:53:33 +0300 Subject: [OpenSIPS-Users] how to debug many dialogs stuck in state 5? In-Reply-To: References: <4de7d18a-9ac5-4b2e-b6df-33d438ec2cb2@opensips.org> Message-ID: Hi, In the same OpenSIPS instance/script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 02.04.2024 15:57, M S wrote: > Hi Bogdan, > When you say at the same time, you mean in the same script? or same > route? or same block for example? > Also what about using dialog and modules that use b2b in underlying > layers, for example mediaexchange? > > Thank you! > > On Mon, Apr 1, 2024 at 10:28 AM Bogdan-Andrei Iancu > wrote: > > You should never use both dialog and b2b modules in the same time, > for the same calls. Trying to have OpenSIPS both Proxy and B2B is > a clear recipe for disaster. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 30.03.2024 07:53, Babak Yakhchali wrote: >> sorry for the late reply. After disabling modules and simplifying >> script logic I found that the issue was b2b entities module being >> enabled, downgrading to 3.2 solved the problem >> >> On Wed, Mar 20, 2024 at 5:19 PM Bogdan-Andrei Iancu >> wrote: >> >> Hi, >> >> What OpenSIPS version are you using? and what module do you >> use on top >> of the dialog module? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 13.03.2024 13:55, Babak Yakhchali wrote: >> > Hi >> > When calling and immediately cancelling, the call is ended but >> > dlg_list shows the dialog stuck in state 5. Increasing log >> level to 4 >> > shows these messages: >> > Mar 13 15:15:37 : DBG:tm:timer_routine: timer >> > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26 >> > Mar 13 15:15:37 : DBG:tm:delete_handler: removing >> 0x7f1c861d0630 >> > Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell >> 0x7f1c861d0630: >> > can't delete -- still reffed (-1) >> > Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2 >> > Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]: >> 0x7f1c861d06e0 (28) >> > Mar 13 15:15:37 : DBG:tm:delete_handler: done >> > Mar 13 15:15:39 : DBG:tm:timer_routine: timer >> > routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28 >> > Mar 13 15:15:39 : DBG:tm:delete_handler: removing >> 0x7f1c861d0630 >> > Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell >> 0x7f1c861d0630: >> > can't delete -- still reffed (-1) >> > Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2 >> > Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]: >> 0x7f1c861d06e0 (30) >> > Mar 13 15:15:39 : DBG:tm:delete_handler: done >> > >> > How can I debug the issue? What are the possible causes of >> this? >> > thanks >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Fri Apr 5 09:38:13 2024 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Fri, 05 Apr 2024 11:38:13 +0200 Subject: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed In-Reply-To: <80e7c4bf-a581-4c4e-b899-d37a1cbf3618@opensips.org> References: <84BD48FD-8F95-4798-8201-7F87E54DAE4C@free.fr> <858EE736-AF3A-46A4-8EBC-8ABF8BC30A8E@free.fr> <027f24dd-f225-468b-9d05-ce31e3a3ee1c@opensips.org> <14EC8DDF-6F01-4C71-91D5-F0A567CD23C9@free.fr> <80e7c4bf-a581-4c4e-b899-d37a1cbf3618@opensips.org> Message-ID: Ho Bogdan, You are right, I implemented the recommendation successfully. Thanks De : Bogdan-Andrei Iancu Date : jeudi 21 mars 2024 à 18:06 À : Alain Bieuzent , OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed HI Alan, I suspect a scripting issue, making the received hop-by-hop ACK (for negative replies) to be "handled" by the dialog module, which treats it as a end-2-end ACK (for 200 OK reply). I guess your script lacks this block https://github.com/OpenSIPS/opensips/blob/master/etc/opensips.cfg#L112 when comes to sequential requests; this block needs to be before the loose_route or any dialog/TH matching. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer   https://www.opensips-solutions.com   https://www.siphub.com On 21.03.2024 12:29, Alain Bieuzent wrote: Hi bogdan, Find attached the sip traces where we have this error log: Mar 21 11:24:13 lbsip-rtpe-test opensips[3108]: WARNING:dialog:dlg_onroute: tight matching failed for ACK with callid='6a543fbd6eabdc7b797dfd6d56db1568 at 10.101.180.176:5060'/52, ftag='as5625de52'/10, ttag='b5a7-94b8b0d1dd76318f3d174a911be4ece2'/37 and direction=1 Mar 21 11:24:13 lbsip-rtpe-test opensips[3108]: WARNING:dialog:dlg_onroute: dialog identification elements are callid='6a543fbd6eabdc7b797dfd6d56db1568 at 10.101.180.176:5060'/52, caller tag='as5625de52'/10, callee tag='3920005391-2017127892'/21 De : Bogdan-Andrei Iancu Date : mercredi 20 mars 2024 à 16:45 À : Alain Bieuzent , OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed Not the logs, but the trace/ SIP capture - I need to see the sip traffic to understand the context of the issue. Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 17:14, Alain Bieuzent wrote: Bellow the link to the traces : https://www.transfernow.net/dl/20240320jzEA98Vz De : Bogdan-Andrei Iancu Date : mercredi 20 mars 2024 à 14:29 À : OpenSIPS users mailling list , Alain Bieuzent Objet : Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed Hi, The script snippet has nothing to do with the matching - it is about a 200-ok ACK to be matched by the dialog module and the TO-tag does not match at all. The link to the trace expired, if you could repost. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 12.03.2024 14:57, Alain Bieuzent wrote: Hi all, I am in the process of migrating my proxies from 3.3.9 to 3.4.4. In 3.4 I find myself with a fault when I try to rewrite a final cause. if (t_check_status("500")) { send_reply(503,"Service Unavailable"); exit; } with 3.3 version, this code works perfectly, since 3.4 I have this error: WARNING:dialog:dlg_onroute: tight matching failed for ACK with callid='67457c401e9ea52e2750100979253a7a at 10.101.180.177:5060'/52, ftag='as5328cea7'/10, ttag='b5a7-2a346bb9d510da893a ee7cda850584cb'/37 and direction=1 WARNING:dialog:dlg_onroute: dialog identification elements are callid='67457c401e9ea52e2750100979253a7a at 10.101.180.177:5060'/52, caller tag='as5328cea7'/10, callee tag='3919223738-770090972'/ 20 Has anyone encountered this error before? Complete traces here : https://www.transfernow.net/dl/20240312qvAFfIAL. Regards _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Fri Apr 5 09:46:11 2024 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Fri, 05 Apr 2024 11:46:11 +0200 Subject: [OpenSIPS-Users] log_next_state_dlg: bogus event 8 in state 5 Message-ID: <8F677B45-28CE-42A1-B4BB-D5125B22689C@free.fr> Hi all, I have some logs like this : WARNING:dialog:log_next_state_dlg: bogus event 8 in state 5 for dlg this case happens when I received a REINVITE from provider just after a BYE from customer, but not yest confirmed by a 200OK. I added this part of code to reply with a 481 if (topology_hiding_match()) {                 if (is_method("INVITE"))                         {                                 if ($DLG_status == 5 ) ### INVITE received for an ended dialog                                 {                                         sl_send_reply(481,"Call/Transaction Does Not Exist");                                         exit;                                 } } The code works, but I continue to receive these error messages in the logs, how can I delete them (without changing the log level)? THANKS               -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Fri Apr 5 14:06:52 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 5 Apr 2024 14:06:52 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: If you are executing that as the first command in the main route block the dialog will not have been created yet. The $DLG_status variable is only valid after the dialog has been created either by a call to create_dialog or by using another module that requires and auto-creates a dialog. Additionally, $DLG_status is documented to only be available for sequential requests, and only after calling loose_route (or match_dialog or topology_hiding_match, I believe). https://opensips.org/docs/modules/3.4.x/dialog.html#pv_DLG_status https://opensips.org/docs/modules/3.4.x/dialog.html#func_create_dialog I still don’t understand why you need some event to capture the start of the call. Even if you are not creating the dialog directly, your routing script has to have some sort of logic that is specific to a new call. Whatever you are trying to “capture” can be captured there. If you cannot identify the path a new call will take through your script, I don’t think anyone here will be able to help you much, at least not without the entire script. Ben Newlin From: Users on behalf of Prathibha B Date: Friday, April 5, 2024 at 3:42 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ route { if($DLG_status == 1) xlog("UNCONFIRMED\n"); } On Fri, 5 Apr 2024 at 13:09, Prathibha B > wrote: I am not getting $DLG_status = UNCONFORMED at the start of the call. Can someone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 6 01:44:21 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 01:44:21 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: If someone attempts to make a call and cancels it, I need to maintain a log of those calls for reporting purposes. Sent from Outlook for Android ________________________________ From: Users on behalf of Ben Newlin Sent: Friday, April 5, 2024 7:36:52 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications If you are executing that as the first command in the main route block the dialog will not have been created yet. The $DLG_status variable is only valid after the dialog has been created either by a call to create_dialog or by using another module that requires and auto-creates a dialog. Additionally, $DLG_status is documented to only be available for sequential requests, and only after calling loose_route (or match_dialog or topology_hiding_match, I believe). https://opensips.org/docs/modules/3.4.x/dialog.html#pv_DLG_status https://opensips.org/docs/modules/3.4.x/dialog.html#func_create_dialog I still don’t understand why you need some event to capture the start of the call. Even if you are not creating the dialog directly, your routing script has to have some sort of logic that is specific to a new call. Whatever you are trying to “capture” can be captured there. If you cannot identify the path a new call will take through your script, I don’t think anyone here will be able to help you much, at least not without the entire script. Ben Newlin From: Users on behalf of Prathibha B Date: Friday, April 5, 2024 at 3:42 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ route { if($DLG_status == 1) xlog("UNCONFIRMED\n"); } On Fri, 5 Apr 2024 at 13:09, Prathibha B > wrote: I am not getting $DLG_status = UNCONFORMED at the start of the call. Can someone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 6 05:06:05 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 10:36:05 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: PFA On Sat, 6 Apr 2024 at 07:14, Prathibha B wrote: > If someone attempts to make a call and cancels it, I need to maintain a > log of those calls for reporting purposes. > > Sent from Outlook for Android > ------------------------------ > *From:* Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Sent:* Friday, April 5, 2024 7:36:52 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > > If you are executing that as the first command in the main route block the > dialog will not have been created yet. The $DLG_status variable is only > valid after the dialog has been created either by a call to create_dialog > or by using another module that requires and auto-creates a dialog. > > > > Additionally, $DLG_status is documented to only be available for > sequential requests, and only after calling loose_route (or match_dialog or > topology_hiding_match, I believe). > > > > https://opensips.org/docs/modules/3.4.x/dialog.html#pv_DLG_status > > https://opensips.org/docs/modules/3.4.x/dialog.html#func_create_dialog > > > > I still don’t understand why you need some event to capture the start of > the call. Even if you are not creating the dialog directly, your routing > script has to have some sort of logic that is specific to a new call. > Whatever you are trying to “capture” can be captured there. > > > > If you cannot identify the path a new call will take through your script, > I don’t think anyone here will be able to help you much, at least not > without the entire script. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Friday, April 5, 2024 at 3:42 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > route { > if($DLG_status == 1) > xlog("UNCONFIRMED\n"); > > } > > > > On Fri, 5 Apr 2024 at 13:09, Prathibha B wrote: > > I am not getting $DLG_status = UNCONFORMED at the start of the call. Can > someone help? > > > > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips-cc.cfg Type: application/octet-stream Size: 21287 bytes Desc: not available URL: From prathibhab.tvm at gmail.com Sat Apr 6 07:34:18 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 07:34:18 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: When will the dialog status be in unconfirmed state? I've checked it the initial state of the call before getting any response. It returns Early state if the state is trying. But never returns unconfirmed state. Sent from Outlook for Android ________________________________ From: Prathibha B Sent: Saturday, April 6, 2024 10:36:05 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications PFA On Sat, 6 Apr 2024 at 07:14, Prathibha B > wrote: If someone attempts to make a call and cancels it, I need to maintain a log of those calls for reporting purposes. Sent from Outlook for Android ________________________________ From: Users > on behalf of Ben Newlin > Sent: Friday, April 5, 2024 7:36:52 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications If you are executing that as the first command in the main route block the dialog will not have been created yet. The $DLG_status variable is only valid after the dialog has been created either by a call to create_dialog or by using another module that requires and auto-creates a dialog. Additionally, $DLG_status is documented to only be available for sequential requests, and only after calling loose_route (or match_dialog or topology_hiding_match, I believe). https://opensips.org/docs/modules/3.4.x/dialog.html#pv_DLG_status https://opensips.org/docs/modules/3.4.x/dialog.html#func_create_dialog I still don’t understand why you need some event to capture the start of the call. Even if you are not creating the dialog directly, your routing script has to have some sort of logic that is specific to a new call. Whatever you are trying to “capture” can be captured there. If you cannot identify the path a new call will take through your script, I don’t think anyone here will be able to help you much, at least not without the entire script. Ben Newlin From: Users > on behalf of Prathibha B > Date: Friday, April 5, 2024 at 3:42 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ route { if($DLG_status == 1) xlog("UNCONFIRMED\n"); } On Fri, 5 Apr 2024 at 13:09, Prathibha B > wrote: I am not getting $DLG_status = UNCONFORMED at the start of the call. Can someone help? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 6 09:29:45 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 14:59:45 +0530 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: will accounting module help here? On Sat, 6 Apr 2024 at 13:04, Prathibha B wrote: > When will the dialog status be in unconfirmed state? I've checked it the > initial state of the call before getting any response. It returns Early > state if the state is trying. But never returns unconfirmed state. > > Sent from Outlook for Android > ------------------------------ > *From:* Prathibha B > *Sent:* Saturday, April 6, 2024 10:36:05 AM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > PFA > > On Sat, 6 Apr 2024 at 07:14, Prathibha B wrote: > > If someone attempts to make a call and cancels it, I need to maintain a > log of those calls for reporting purposes. > > Sent from Outlook for Android > ------------------------------ > *From:* Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Sent:* Friday, April 5, 2024 7:36:52 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] external applications > > > If you are executing that as the first command in the main route block the > dialog will not have been created yet. The $DLG_status variable is only > valid after the dialog has been created either by a call to create_dialog > or by using another module that requires and auto-creates a dialog. > > > > Additionally, $DLG_status is documented to only be available for > sequential requests, and only after calling loose_route (or match_dialog or > topology_hiding_match, I believe). > > > > https://opensips.org/docs/modules/3.4.x/dialog.html#pv_DLG_status > > https://opensips.org/docs/modules/3.4.x/dialog.html#func_create_dialog > > > > I still don’t understand why you need some event to capture the start of > the call. Even if you are not creating the dialog directly, your routing > script has to have some sort of logic that is specific to a new call. > Whatever you are trying to “capture” can be captured there. > > > > If you cannot identify the path a new call will take through your script, > I don’t think anyone here will be able to help you much, at least not > without the entire script. > > > > Ben Newlin > > > > *From: *Users on behalf of Prathibha B > > *Date: *Friday, April 5, 2024 at 3:42 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > route { > if($DLG_status == 1) > xlog("UNCONFIRMED\n"); > > } > > > > On Fri, 5 Apr 2024 at 13:09, Prathibha B wrote: > > I am not getting $DLG_status = UNCONFORMED at the start of the call. Can > someone help? > > > > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 6 14:35:44 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 14:35:44 +0000 Subject: [OpenSIPS-Users] Continously ringing Message-ID: User A makes a call to user B. User B didn't accept the call. User B stopped ringing after 1 minute. But User A continues to be in the ringing state until it is manually canceled. Sent from Outlook for Android -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 6 14:39:51 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 6 Apr 2024 14:39:51 +0000 Subject: [OpenSIPS-Users] Packet analysis using wireshark Message-ID: I am unable to see the Voip calls in wireshark. For signaling opensips is used. The calls are encrypted and it is webrtc communication. Sent from Outlook for Android -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Sat Apr 6 17:57:56 2024 From: abalashov at evaristesys.com (Alex Balashov) Date: Sat, 6 Apr 2024 13:57:56 -0400 Subject: [OpenSIPS-Users] Continously ringing In-Reply-To: References: Message-ID: <71E8DEF5-B3A1-4655-84D8-39F16150244D@evaristesys.com> Is there a predicate to this sentence? > On Apr 6, 2024, at 10:35 AM, Prathibha B wrote: > > User A makes a call to user B. User B didn't accept the call. User B stopped ringing after 1 minute. But User A continues to be in the ringing state until it is manually canceled. > > Sent from Outlook for Android > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 From abalashov at evaristesys.com Sat Apr 6 17:58:06 2024 From: abalashov at evaristesys.com (Alex Balashov) Date: Sat, 6 Apr 2024 13:58:06 -0400 Subject: [OpenSIPS-Users] Packet analysis using wireshark In-Reply-To: References: Message-ID: <5E85D4CE-C30C-4008-B2A9-680908B4BDD1@evaristesys.com> This is known as "working as intended". > On Apr 6, 2024, at 10:39 AM, Prathibha B wrote: > > I am unable to see the Voip calls in wireshark. For signaling opensips is used. The calls are encrypted and it is webrtc communication. > > Sent from Outlook for Android > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 From liviu at opensips.org Mon Apr 8 10:07:43 2024 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 8 Apr 2024 13:07:43 +0300 Subject: [OpenSIPS-Users] Continously ringing In-Reply-To: References: Message-ID: <748e72fd-531b-4aca-b18f-4f83739f5d65@opensips.org> Make a SIP trace with *sngrep *program and check the routing of the 486 Busy / 603 Decline message sent by phone B.  If A is still in ringing, it most likely means the reply isn't reaching it. It is possible phone A is behind a NAT and its /Via/ SIP header needs a small adjustment for the replies to be properly routed back, using this useful /opensips.cfg/ function call: force_rport() Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events On 06.04.2024 17:35, Prathibha B wrote: > User A makes a call to user B. User B didn't accept the call. User B > stopped ringing after 1 minute. But User A continues to be in the > ringing state until it is manually canceled. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Apr 8 10:14:42 2024 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 8 Apr 2024 13:14:42 +0300 Subject: [OpenSIPS-Users] Packet analysis using wireshark In-Reply-To: References: Message-ID: <6a5b5fe6-f46f-4add-a43d-a77e8e25fe2a@opensips.org> If you are not able to decode the WebRTC TLS connection in Wireshark, it's possible you are dealing with a TLS 1.3 connection. In TLS 1.3, there is an extra "secrets" file which must be plugged into Wireshark before it can decode the communication, which contains transient data (per connection!).  It is no longer sufficient to go to Edit -> Preferences -> Protocols -> TLS / SSL -> *RSA keys list* and plug in your private key.  In that same dialog box, the field *(Pre)-Master-Secret log filename* also becomes mandatory. Now, how to obtain the Master-Secret file?  In Chrome/Firefox as well as in cURL, you should find support for the *SSLKEYLOGFILE=* environment variable. Just make sure to set this variable to the desired filepath before running the WebRTC client and it /should/ dump the secrets there.  Which will ultimately get picked up by Wireshark and the traffic will decode. Good luck! :) Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events On 06.04.2024 17:39, Prathibha B wrote: > I am unable to see the Voip calls in wireshark. For signaling opensips > is used. The calls are encrypted and it is webrtc communication. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Apr 8 10:23:29 2024 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 8 Apr 2024 13:23:29 +0300 Subject: [OpenSIPS-Users] log_next_state_dlg: bogus event 8 in state 5 In-Reply-To: <8F677B45-28CE-42A1-B4BB-D5125B22689C@free.fr> References: <8F677B45-28CE-42A1-B4BB-D5125B22689C@free.fr> Message-ID: <23c78662-9f86-4c9d-b7c9-d4dbee30f5ff@opensips.org> Hi Alain, Event 8 stands for DLG_EVENT_REQ in the C code, which means: "mid-dialog request". I would recommend you don't turn off that warning message, because there may be several other events (5? 7? etc.) which could reach the same codepath and could be useful hints for troubleshooting.  Rather, try to make a /monitoring exception/ for the specific "event 8 state 5" warning message in order to avoid unwanted alerts. PS: as far as the function itself goes, there is no way of turning the message off, only by raising the /log_level/ -- again, not recommended (warnings are useful). Best regards, Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events On 05.04.2024 12:46, Alain Bieuzent wrote: > > WARNING:dialog:log_next_state_dlg: bogus event 8 in state 5 for dlg > > The code works, but I continue to receive these error messages in the > logs, how can I delete them (without changing the log level)? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 8 10:32:18 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 8 Apr 2024 16:02:18 +0530 Subject: [OpenSIPS-Users] Packet analysis using wireshark In-Reply-To: <6a5b5fe6-f46f-4add-a43d-a77e8e25fe2a@opensips.org> References: <6a5b5fe6-f46f-4add-a43d-a77e8e25fe2a@opensips.org> Message-ID: Thank you. On Mon, 8 Apr 2024 at 15:44, Liviu Chircu wrote: > If you are not able to decode the WebRTC TLS connection in Wireshark, it's > possible you are dealing with a TLS 1.3 connection. > > In TLS 1.3, there is an extra "secrets" file which must be plugged into > Wireshark before it can decode the communication, which contains transient > data (per connection!). It is no longer sufficient to go to Edit -> > Preferences -> Protocols -> TLS / SSL -> *RSA keys list* and plug in your > private key. In that same dialog box, the field *(Pre)-Master-Secret log > filename* also becomes mandatory. > > Now, how to obtain the Master-Secret file? In Chrome/Firefox as well as > in cURL, you should find support for the *SSLKEYLOGFILE=* environment > variable. Just make sure to set this variable to the desired filepath > before running the WebRTC client and it *should* dump the secrets there. > Which will ultimately get picked up by Wireshark and the traffic will > decode. > > Good luck! :) > > Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS Summit 2024 Valencia, May 14-17 | www.opensips.org/events > > On 06.04.2024 17:39, Prathibha B wrote: > > I am unable to see the Voip calls in wireshark. For signaling opensips is > used. The calls are encrypted and it is webrtc communication. > > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Mon Apr 8 11:52:18 2024 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Mon, 08 Apr 2024 13:52:18 +0200 Subject: [OpenSIPS-Users] log_next_state_dlg: bogus event 8 in state 5 In-Reply-To: <23c78662-9f86-4c9d-b7c9-d4dbee30f5ff@opensips.org> References: <8F677B45-28CE-42A1-B4BB-D5125B22689C@free.fr> <23c78662-9f86-4c9d-b7c9-d4dbee30f5ff@opensips.org> Message-ID: <97B75F1C-9F1A-4E90-B059-EB4C0A58E98F@free.fr> Thanks Liviu, See you in valencia ! De : Liviu Chircu Date : lundi 8 avril 2024 à 12:23 À : OpenSIPS users mailling list , Alain Bieuzent Objet : Re: [OpenSIPS-Users] log_next_state_dlg: bogus event 8 in state 5 Hi Alain, Event 8 stands for DLG_EVENT_REQ in the C code, which means: "mid-dialog request". I would recommend you don't turn off that warning message, because there may be several other events (5? 7? etc.) which could reach the same codepath and could be useful hints for troubleshooting. Rather, try to make a monitoring exception for the specific "event 8 state 5" warning message in order to avoid unwanted alerts. PS: as far as the function itself goes, there is no way of turning the message off, only by raising the log_level -- again, not recommended (warnings are useful). Best regards, Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit 2024 Valencia, May 14-17 | www.opensips.org/events On 05.04.2024 12:46, Alain Bieuzent wrote: WARNING:dialog:log_next_state_dlg: bogus event 8 in state 5 for dlg The code works, but I continue to receive these error messages in the logs, how can I delete them (without changing the log level)? -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Tue Apr 9 09:29:01 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Tue, 9 Apr 2024 14:59:01 +0530 Subject: [OpenSIPS-Users] Waiting for 200 OK Message-ID: Hello, I have a requirement where If I get 183 responses then I need to wait for a specific period of time for 200 OK and if 200 OK is not received in that time then I need to send Reinvte back to UA with some modifications. I have gone through the SST module but have not found anything relevant to match my requirements. Can anyone suggest some module or function on which I can go further to meet my requirements. Regards, *Parthesh Bhavsar | Software Engineer | VOIP* -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 9 14:06:16 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 9 Apr 2024 14:06:16 +0000 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: Message-ID: The timing requirement can be solved using the $T_fr_timeout parameter [1]. For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout Ben Newlin From: Users on behalf of Parthesh Bhavsar via Users Date: Tuesday, April 9, 2024 at 5:31 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Waiting for 200 OK EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello, I have a requirement where If I get 183 responses then I need to wait for a specific period of time for 200 OK and if 200 OK is not received in that time then I need to send Reinvte back to UA with some modifications. I have gone through the SST module but have not found anything relevant to match my requirements. Can anyone suggest some module or function on which I can go further to meet my requirements. Regards, Parthesh Bhavsar | Software Engineer | VOIP [https://ecosmobnew.ecosmob.net/wp-content/uploads/2024/04/opensip-email-signature.jpg] Disclaimer In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. Confidentiality This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. Caution for viruses, malware etc. This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Tue Apr 9 14:35:25 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Tue, 9 Apr 2024 20:05:25 +0530 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: Message-ID: It seems after setting T_fr_timeout parameter it was sending a CANCEL request to another leg but for my requirement I need to send Reinvite. Also for generating Reinvite I have used dlg_send_sequential() as I use opensips as a proxy server. Any other modules on which I look for? Regards, *Parthesh Bhavsar | Software Engineer | VOIP* On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin wrote: > The timing requirement can be solved using the $T_fr_timeout parameter [1]. > > > > For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. > > > > [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout > > > > Ben Newlin > > > > *From: *Users on behalf of Parthesh > Bhavsar via Users > *Date: *Tuesday, April 9, 2024 at 5:31 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hello, > > I have a requirement where If I get 183 responses then I need to wait for > a specific period of time for 200 OK and if 200 OK is not received in that > time then I need to send Reinvte back to UA with some modifications. I have > gone through the SST module but have not found anything relevant to match > my requirements. Can anyone suggest some module or function on which I can > go further to meet my requirements. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > > > * * > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > > > *Confidentiality* > > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 9 14:44:52 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 9 Apr 2024 14:44:52 +0000 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: Message-ID: You’re right, that timer would drop the call. Not sure why I was thinking that would work. Sorry! Ben Newlin From: Parthesh Bhavsar Date: Tuesday, April 9, 2024 at 10:35 AM To: Ben Newlin Cc: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Waiting for 200 OK EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ It seems after setting T_fr_timeout parameter it was sending a CANCEL request to another leg but for my requirement I need to send Reinvite. Also for generating Reinvite I have used dlg_send_sequential() as I use opensips as a proxy server. Any other modules on which I look for? Regards, Parthesh Bhavsar | Software Engineer | VOIP On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin > wrote: The timing requirement can be solved using the $T_fr_timeout parameter [1]. For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout Ben Newlin From: Users > on behalf of Parthesh Bhavsar via Users > Date: Tuesday, April 9, 2024 at 5:31 AM To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] Waiting for 200 OK EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello, I have a requirement where If I get 183 responses then I need to wait for a specific period of time for 200 OK and if 200 OK is not received in that time then I need to send Reinvte back to UA with some modifications. I have gone through the SST module but have not found anything relevant to match my requirements. Can anyone suggest some module or function on which I can go further to meet my requirements. Regards, Parthesh Bhavsar | Software Engineer | VOIP [https://ecosmobnew.ecosmob.net/wp-content/uploads/2024/04/opensip-email-signature.jpg] Disclaimer In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. Confidentiality This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. Caution for viruses, malware etc. This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. [https://ecosmobnew.ecosmob.net/wp-content/uploads/2024/04/opensip-email-signature.jpg] Disclaimer In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. Confidentiality This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. Caution for viruses, malware etc. This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. -------------- next part -------------- An HTML attachment was scrubbed... URL: From medeanwz at gmail.com Tue Apr 9 14:53:27 2024 From: medeanwz at gmail.com (M S) Date: Tue, 9 Apr 2024 16:53:27 +0200 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: Message-ID: Maybe using a timer route is the solution? Save 183 time in cache and check if 200 ok is received within desired period, otherwise send reinvite... On Tue, Apr 9, 2024, 4:48 PM Ben Newlin wrote: > You’re right, that timer would drop the call. Not sure why I was thinking > that would work. Sorry! > > > > Ben Newlin > > > > *From: *Parthesh Bhavsar > *Date: *Tuesday, April 9, 2024 at 10:35 AM > *To: *Ben Newlin > *Cc: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > It seems after setting T_fr_timeout parameter it was sending a CANCEL > request to another leg but for my requirement I need to send Reinvite. Also > for generating Reinvite I have used dlg_send_sequential() as I use > opensips as a proxy server. Any other modules on which I look for? > > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > > > On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin wrote: > > The timing requirement can be solved using the $T_fr_timeout parameter [1]. > > > > For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. > > > > [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout > > > > Ben Newlin > > > > *From: *Users on behalf of Parthesh > Bhavsar via Users > *Date: *Tuesday, April 9, 2024 at 5:31 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hello, > > I have a requirement where If I get 183 responses then I need to wait for > a specific period of time for 200 OK and if 200 OK is not received in that > time then I need to send Reinvte back to UA with some modifications. I have > gone through the SST module but have not found anything relevant to match > my requirements. Can anyone suggest some module or function on which I can > go further to meet my requirements. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > > > * * > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > > > *Confidentiality* > > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > > > > * * > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > > > *Confidentiality* > > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 9 14:55:27 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 9 Apr 2024 17:55:27 +0300 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: Message-ID: <74d1f65e-9d60-4029-9a04-000d70ce6450@opensips.org> Hi, my 2 cents here - how comes you want to send a RE-INVITE _BEFORE_ having the dialog established??? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 09.04.2024 17:35, Parthesh Bhavsar via Users wrote: > It seems after setting T_fr_timeout parameter it was sending a CANCEL > request to another leg but for my requirement I need to send Reinvite. > Also for generating Reinvite I have used dlg_send_sequential() as I > use opensips as a proxy server. Any other modules on which I look for? > Regards, > *Parthesh Bhavsar | Software Engineer | VOIP* > > > On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin wrote: > > The timing requirement can be solved using the $T_fr_timeout > parameter [1]. > > For sending a re-Invite back to the UAC I believe you’d have to be > a B2BUA. > > [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout > > Ben Newlin > > *From: *Users on behalf of > Parthesh Bhavsar via Users > *Date: *Tuesday, April 9, 2024 at 5:31 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > ------------------------------------------------------------------------ > > Hello, > > I have a requirement where If I get 183 responses then I need to > wait for a specific period of time for 200 OK and if 200 OK is not > received in that time then I need to send Reinvte back to UA with > some modifications. I have gone through the SST module but have > not  found anything relevant to match my requirements. Can anyone > suggest some module or function on which I can go further to meet > my requirements. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > ** > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our > website, any views or opinions presented in this email are solely > those of the originator and do not necessarily represent those of > the Company or its sister concerns. Any liability (in negligence, > contract or otherwise) arising from any third party taking any > action, or refraining from taking any action on the basis of any > of the information contained in this email is hereby excluded. > > *Confidentiality* > > This communication (including any attachment/s) is intended only > for the use of the addressee(s) and contains information that is > PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, > distribution, or copying of this communication is prohibited. > Please inform originator if you have received it in error. > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of > viruses, trojans, similar or new contaminants/malware, > interceptions or interference, and may not be compatible with your > systems. You shall carry out virus/malware scanning on your own > before opening any attachment to this e-mail. The sender of this > e-mail and Company including its sister concerns shall not be > liable for any damage that may incur to you as a result of > viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > > > *https://www.ecosmob.com/opensips-summit/ > > * > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our > website, any views or opinions presented in this email are solely > those of the originator and do not necessarily represent those of the > Company or its sister concerns. Any liability (in negligence, contract > or otherwise) arising from any third party taking any action, or > refraining from taking any action on the basis of any of the > information contained in this email is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for > the use of the addressee(s) and contains information that is > PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, > distribution, or copying of this communication is prohibited. Please > inform originator if you have received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of > viruses, trojans, similar or new contaminants/malware, interceptions > or interference, and may not be compatible with your systems. You > shall carry out virus/malware scanning on your own before opening any > attachment to this e-mail. The sender of this e-mail and Company > including its sister concerns shall not be liable for any damage that > may incur to you as a result of viruses, incompleteness of this > message, a delay in receipt of this message or any other computer > problems. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Apr 9 15:40:50 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 9 Apr 2024 15:40:50 +0000 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: <74d1f65e-9d60-4029-9a04-000d70ce6450@opensips.org> References: <74d1f65e-9d60-4029-9a04-000d70ce6450@opensips.org> Message-ID: This is a valid point that I missed. You can’t send an INVITE back to the UAC while the original INVITE remains unanswered. The UAC should/must respond with a 491 Request Pending. This is the use case that UPDATE was created for, and that would work if the UAC supports it. If it must be INVITE then you would have to be a B2BUA because you would have to answer the initial INVITE from the UAC while the outgoing INVITE was still in progress, so that you could do the re-INVITE. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, April 9, 2024 at 10:55 AM To: Parthesh Bhavsar , OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] Waiting for 200 OK EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi, my 2 cents here - how comes you want to send a RE-INVITE _BEFORE_ having the dialog established??? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 09.04.2024 17:35, Parthesh Bhavsar via Users wrote: It seems after setting T_fr_timeout parameter it was sending a CANCEL request to another leg but for my requirement I need to send Reinvite. Also for generating Reinvite I have used dlg_send_sequential() as I use opensips as a proxy server. Any other modules on which I look for? Regards, Parthesh Bhavsar | Software Engineer | VOIP On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin > wrote: The timing requirement can be solved using the $T_fr_timeout parameter [1]. For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout Ben Newlin From: Users > on behalf of Parthesh Bhavsar via Users > Date: Tuesday, April 9, 2024 at 5:31 AM To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] Waiting for 200 OK EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello, I have a requirement where If I get 183 responses then I need to wait for a specific period of time for 200 OK and if 200 OK is not received in that time then I need to send Reinvte back to UA with some modifications. I have gone through the SST module but have not found anything relevant to match my requirements. Can anyone suggest some module or function on which I can go further to meet my requirements. Regards, Parthesh Bhavsar | Software Engineer | VOIP [https://ecosmobnew.ecosmob.net/wp-content/uploads/2024/04/opensip-email-signature.jpg] Disclaimer In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. Confidentiality This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. Caution for viruses, malware etc. This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. [https://ecosmobnew.ecosmob.net/wp-content/uploads/2024/04/opensip-email-signature.jpg] Disclaimer In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. Confidentiality This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. Caution for viruses, malware etc. This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From John.Sliney at lcs.com Tue Apr 9 17:59:24 2024 From: John.Sliney at lcs.com (John Sliney) Date: Tue, 9 Apr 2024 17:59:24 +0000 Subject: [OpenSIPS-Users] Load Balancer add destination as probing status Message-ID: Is there currently a way to add a destination to the load_balancer set with a status of disabled but with probing enabled, like the dispatcher table 'state' column allows? I have an osips working as a SIP ingress/egress for Kubernetes traffic and as Asterisk Pods are created they are added to the load_balancer table as destinations, but OpenSIPS will route traffic to them before they're fully ready. I've attempted to automatically mark them as disabled with the mi lb_status command and let the probe enable them, but disabling via mi seems to disable probing as well, so they're never actually enabled. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Tue Apr 9 20:01:23 2024 From: Johan at democon.be (Johan De Clercq) Date: Tue, 9 Apr 2024 22:01:23 +0200 Subject: [OpenSIPS-Users] Load Balancer add destination as probing status In-Reply-To: References: Message-ID: Can you tell me how to disable a load balancer destination via mi ? On Tue, 9 Apr 2024, 20:02 John Sliney, wrote: > Is there currently a way to add a destination to the load_balancer set > with a status of disabled but with probing enabled, like the dispatcher > table 'state' column allows? > > I have an osips working as a SIP ingress/egress for Kubernetes traffic > and as Asterisk Pods are created they are added to the load_balancer table > as destinations, but OpenSIPS will route traffic to them before they're > fully ready. > I've attempted to automatically mark them as disabled with the mi > lb_status command and let the probe enable them, but disabling via mi seems > to disable probing as well, so they're never actually enabled. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From John.Sliney at lcs.com Tue Apr 9 20:06:54 2024 From: John.Sliney at lcs.com (John Sliney) Date: Tue, 9 Apr 2024 20:06:54 +0000 Subject: [OpenSIPS-Users] Load Balancer add destination as probing status In-Reply-To: References: Message-ID: Using the lb_status command with two params, the ID of the destination and desired status (0 if you want to disable), this does seem to disable probing though so the destination will stay disabled until you manually set status back to 1 https://opensips.org/docs/modules/3.2.x/load_balancer.html#mi_lb_status ________________________________ From: Users on behalf of Johan De Clercq Sent: Tuesday, April 9, 2024 4:01 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Load Balancer add destination as probing status CAUTION: EXTERNAL Can you tell me how to disable a load balancer destination via mi ? On Tue, 9 Apr 2024, 20:02 John Sliney, > wrote: Is there currently a way to add a destination to the load_balancer set with a status of disabled but with probing enabled, like the dispatcher table 'state' column allows? I have an osips working as a SIP ingress/egress for Kubernetes traffic and as Asterisk Pods are created they are added to the load_balancer table as destinations, but OpenSIPS will route traffic to them before they're fully ready. I've attempted to automatically mark them as disabled with the mi lb_status command and let the probe enable them, but disabling via mi seems to disable probing as well, so they're never actually enabled. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Tue Apr 9 21:21:02 2024 From: rob.dyck at telus.net (Robert Dyck) Date: Tue, 09 Apr 2024 14:21:02 -0700 Subject: [OpenSIPS-Users] Message buffer formatting Message-ID: <4341112.ejJDZkT8p0@leno.mylan> In the past I would insert xlog with $mb into my script for debugging purposes. Now I find that the message buffer output is not being formatted. Instead of Message Buffer REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport I get Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z 9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012 Instead of LF I get #012 Instead of CR I get #015 What may have caused this change and how can I restore the old behaviour? -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Wed Apr 10 06:01:15 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Wed, 10 Apr 2024 11:31:15 +0530 Subject: [OpenSIPS-Users] Waiting for 200 OK In-Reply-To: References: <74d1f65e-9d60-4029-9a04-000d70ce6450@opensips.org> Message-ID: Sorry my bad! Actually I need to Wait for a specific time after 200 OK received from Bleg. So in that specific time If I received Re-invite from Aleg then I need to simply realy or if not received then need to do some other operation. Regards, *Parthesh Bhavsar | Software Engineer | VOIP* On Tue, Apr 9, 2024 at 9:11 PM Ben Newlin wrote: > This is a valid point that I missed. You can’t send an INVITE back to the > UAC while the original INVITE remains unanswered. The UAC should/must > respond with a 491 Request Pending. > > > > This is the use case that UPDATE was created for, and that would work if > the UAC supports it. If it must be INVITE then you would have to be a B2BUA > because you would have to answer the initial INVITE from the UAC while the > outgoing INVITE was still in progress, so that you could do the re-INVITE. > > > > Ben Newlin > > > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, April 9, 2024 at 10:55 AM > *To: *Parthesh Bhavsar , OpenSIPS users > mailling list , Ben Newlin < > Ben.Newlin at genesys.com> > *Subject: *Re: [OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi, > > my 2 cents here - how comes you want to send a RE-INVITE _BEFORE_ having > the dialog established??? > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 09.04.2024 17:35, Parthesh Bhavsar via Users wrote: > > It seems after setting T_fr_timeout parameter it was sending a CANCEL > request to another leg but for my requirement I need to send Reinvite. Also > for generating Reinvite I have used dlg_send_sequential() as I use > opensips as a proxy server. Any other modules on which I look for? > > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > > > On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin wrote: > > The timing requirement can be solved using the $T_fr_timeout parameter [1]. > > > > For sending a re-Invite back to the UAC I believe you’d have to be a B2BUA. > > > > [1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout > > > > Ben Newlin > > > > *From: *Users on behalf of Parthesh > Bhavsar via Users > *Date: *Tuesday, April 9, 2024 at 5:31 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Waiting for 200 OK > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hello, > > I have a requirement where If I get 183 responses then I need to wait for > a specific period of time for 200 OK and if 200 OK is not received in that > time then I need to send Reinvte back to UA with some modifications. I have > gone through the SST module but have not found anything relevant to match > my requirements. Can anyone suggest some module or function on which I can > go further to meet my requirements. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > > > * * > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > > > *Confidentiality* > > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > > > > * * > > *Disclaimer* > > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > > > *Confidentiality* > > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > > > *Caution for viruses, malware etc.* > > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 10 06:47:05 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 10 Apr 2024 09:47:05 +0300 Subject: [OpenSIPS-Users] Message buffer formatting In-Reply-To: <4341112.ejJDZkT8p0@leno.mylan> References: <4341112.ejJDZkT8p0@leno.mylan> Message-ID: <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> There is no formatting added, maybe the diff comes for the actual logging. What are the 2 versions you tested ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10.04.2024 00:21, Robert Dyck wrote: > > In the past I would insert xlog with $mb into my script for debugging > purposes. Now I find that the message buffer output is not being > formatted. > > > Instead of > > > Message Buffer > > REGISTER sip:192.168.1.2 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport > > > I get > > > Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: > SIP/2.0/UDP 192.168.1.4:5070;branch=z > 9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012 > > > Instead of LF I get #012 > > Instead of CR I get #015 > > > What may have caused this change and how can I restore the old behaviour? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From y.kirsanov at gmail.com Wed Apr 10 10:21:28 2024 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Wed, 10 Apr 2024 20:21:28 +1000 Subject: [OpenSIPS-Users] UsrLoc Events are not working with Full-Sharing-CacheDB-Cluster and MongoDB Message-ID: Hi, I've got three OpenSIPS nodes in a cluster and it was working fine in 'full-sharing-cluster' mode. Then I decided to add MongoDB as a cachedb to store all the registrations, so I enabled the mongodb module and added 'cachedb_url' parameter into the usrloc module config. After doing that I noticed that I'm not receiving events for usrloc on all nodes as it was before. For example, registration comes to OpenSIPS node 3, it successfully processes the registration request and stores it into MongoDB. But the other 2 nodes do not see that event! Even if I subscribe for events using following code: t_newtran(); t_wait_for_new_branches(); $avp(filter) = "aor="+$tU; notify_on_event("E_UL_CONTACT_UPDATE",$avp(filter),"fork_call", 20); notify_on_event("E_UL_CONTACT_INSERT",$avp(filter),"fork_call", 20); I'm not receiving the event notifications anywhere but on the node where the SIP REGISTER packet came. This doesn't allow to use 't_inject_branches("event")' command. I have following routes defined for events: event_route[E_UL_CONTACT_INSERT] event_route[E_UL_CONTACT_UPDATE] event_route[E_UL_CONTACT_DELETE] route[fork_call] None of them are getting called on other nodes of the cluster except for the node that accepted the REGISTER packet. Here's my cluster and usrloc configuration: # USRLoc module loadmodule "usrloc.so" modparam("usrloc", "nat_bflag", "NAT") modparam("usrloc", "working_mode_preset", "full-sharing-cachedb-cluster") modparam("usrloc", "use_domain", 0) modparam("usrloc", "location_cluster", 1) modparam("usrloc", "cachedb_url", "mongodb://10.x.x.1,10.x.x.2,10.x.x.3:27017/opensipsDB.userlocation") # Clusterer Module loadmodule "clusterer.so" modparam("clusterer", "db_mode", 0) modparam("clusterer", "my_node_info", "cluster_id=1,node_id=1,url=bin:10.y.y.1:3857,flags=seed") modparam("clusterer", "neighbor_node_info", "cluster_id=1,node_id=2,url=bin:10.y.y.2:3857,flags=seed") modparam("clusterer", "neighbor_node_info", "cluster_id=1,node_id=3,url=bin:10.y.y.3:3857,flags=seed") modparam("clusterer", "seed_fallback_interval", 5) modparam("clusterer", "ping_interval", 1) modparam("clusterer", "ping_timeout", 500) modparam("clusterer", "node_timeout", 10) modparam("clusterer", "sharing_tag", "vip1/1=active") # Event Route module loadmodule "event_route.so" # Event Routing module loadmodule "event_routing.so" # Event Stream loadmodule "event_stream.so" # Registrar module loadmodule "registrar.so" modparam("registrar", "received_avp", "$avp(received)") modparam("registrar", "min_expires", 60) modparam("registrar", "default_expires", 120) modparam("registrar", "max_expires", 3600) modparam("registrar", "max_contacts", 1) modparam("registrar", "attr_avp", "$avp(event_attr)") Am I doing something wrong? How to enable event notifications on all nodes in a full sharing cachedb cluster? Thanks. Best regards, Yury. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Apr 10 13:41:54 2024 From: Johan at democon.be (Johan De Clercq) Date: Wed, 10 Apr 2024 15:41:54 +0200 Subject: [OpenSIPS-Users] on from and to header. Message-ID: This is more a protocol question then anything else. A sends a call B INVITE B at 5.6.7.8 from: ;tag=1 to: call-id: 1 ...... B responds with from and to in e164 format 200 OK from: ;tag=1 to: ; tag=2 call-id: 1 My gut feeling says to me that A will not like this and hence that B should respong with A and B instead of +32A and +32B. But can somebody explain me why or why not? Best regards, Johan. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 10 14:04:19 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 10 Apr 2024 14:04:19 +0000 Subject: [OpenSIPS-Users] on from and to header. In-Reply-To: References: Message-ID: RFC 3261 is explicit that the From and To headers in a response MUST be equal to those in the request, with the exception of the potential addition of a tag in the To header if not already present. https://www.rfc-editor.org/rfc/rfc3261.html#section-8.2.6.2 Ben Newlin From: Users on behalf of Johan De Clercq Date: Wednesday, April 10, 2024 at 9:44 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] on from and to header. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ This is more a protocol question then anything else. A sends a call B INVITE B at 5.6.7.8 from: >;tag=1 to: > call-id: 1 ...... B responds with from and to in e164 format 200 OK from: >;tag=1 to: >; tag=2 call-id: 1 My gut feeling says to me that A will not like this and hence that B should respong with A and B instead of +32A and +32B. But can somebody explain me why or why not? Best regards, Johan. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Wed Apr 10 16:00:09 2024 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 10 Apr 2024 09:00:09 -0700 Subject: [OpenSIPS-Users] Message buffer formatting In-Reply-To: <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> References: <4341112.ejJDZkT8p0@leno.mylan> <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> Message-ID: <9313673.rMLUfLXkoz@leno.mylan> I had been running 3.4. I fired up 3.3. Had to modify the config because syslog syntax had changed. $mb worked as expected. I went back to 3.4. I got notices about the log stuff being deprecated so I went back to the new syntax. Now $mb works as expected on 3.4. I have concluded that something about the original 3.4 config was messed up. All good now. On Tuesday, April 9, 2024 11:47:05 P.M. PDT Bogdan-Andrei Iancu wrote: > There is no formatting added, maybe the diff comes for the actual > logging. What are the 2 versions you tested ? > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 10.04.2024 00:21, Robert Dyck wrote: > > In the past I would insert xlog with $mb into my script for debugging > > purposes. Now I find that the message buffer output is not being > > formatted. > > > > > > Instead of > > > > > > Message Buffer > > > > REGISTER sip:192.168.1.2 SIP/2.0 > > > > Via: SIP/2.0/UDP > > 192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport > > > > > > I get > > > > > > Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: > > SIP/2.0/UDP 192.168.1.4:5070;branch=z > > 9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012 > > > > > > Instead of LF I get #012 > > > > Instead of CR I get #015 > > > > > > What may have caused this change and how can I restore the old behaviour? > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From rob.dyck at telus.net Wed Apr 10 16:11:03 2024 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 10 Apr 2024 09:11:03 -0700 Subject: [OpenSIPS-Users] Message buffer formatting In-Reply-To: <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> References: <4341112.ejJDZkT8p0@leno.mylan> <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> Message-ID: <4984138.0VBMTVartN@leno.mylan> I spoke too soon. When I run opensips in debug mode on the terminal the formatting looks good. If I monitor the log facility the $mb dump is not formatted. Perhaps this is normal? On Tuesday, April 9, 2024 11:47:05 P.M. PDT Bogdan-Andrei Iancu wrote: > There is no formatting added, maybe the diff comes for the actual > logging. What are the 2 versions you tested ? > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 10.04.2024 00:21, Robert Dyck wrote: > > In the past I would insert xlog with $mb into my script for debugging > > purposes. Now I find that the message buffer output is not being > > formatted. > > > > > > Instead of > > > > > > Message Buffer > > > > REGISTER sip:192.168.1.2 SIP/2.0 > > > > Via: SIP/2.0/UDP > > 192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport > > > > > > I get > > > > > > Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: > > SIP/2.0/UDP 192.168.1.4:5070;branch=z > > 9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012 > > > > > > Instead of LF I get #012 > > > > Instead of CR I get #015 > > > > > > What may have caused this change and how can I restore the old behaviour? > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Apr 11 06:08:32 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 11 Apr 2024 09:08:32 +0300 Subject: [OpenSIPS-Users] Message buffer formatting In-Reply-To: <4984138.0VBMTVartN@leno.mylan> References: <4341112.ejJDZkT8p0@leno.mylan> <43122a18-ab4c-4ba3-a88c-db062e48e17d@opensips.org> <4984138.0VBMTVartN@leno.mylan> Message-ID: <32c495d1-adaa-4003-937b-9d0a0782dbed@opensips.org> It sounds like something particular to the way you do the monitoring of the logs. Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10.04.2024 19:11, Robert Dyck wrote: > I spoke too soon. When I run opensips in debug mode on the terminal the > formatting looks good. If I monitor the log facility the $mb dump is not > formatted. Perhaps this is normal? > > On Tuesday, April 9, 2024 11:47:05 P.M. PDT Bogdan-Andrei Iancu wrote: >> There is no formatting added, maybe the diff comes for the actual >> logging. What are the 2 versions you tested ? >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 10.04.2024 00:21, Robert Dyck wrote: >>> In the past I would insert xlog with $mb into my script for debugging >>> purposes. Now I find that the message buffer output is not being >>> formatted. >>> >>> >>> Instead of >>> >>> >>> Message Buffer >>> >>> REGISTER sip:192.168.1.2 SIP/2.0 >>> >>> Via: SIP/2.0/UDP >>> 192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport >>> >>> >>> I get >>> >>> >>> Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: >>> SIP/2.0/UDP 192.168.1.4:5070;branch=z >>> 9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012 >>> >>> >>> Instead of LF I get #012 >>> >>> Instead of CR I get #015 >>> >>> >>> What may have caused this change and how can I restore the old behaviour? >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > From prathibhab.tvm at gmail.com Thu Apr 11 08:57:41 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 14:27:41 +0530 Subject: [OpenSIPS-Users] Call Center error Message-ID: Getting the following error in call center module ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: ERROR:call_center:w_handle_call: failed to set new destination for call Call center code in opensips.cfg: if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { if (!cc_handle_call("support")) { send_reply(403,"Cannot handle call"); exit; } } -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 08:59:41 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 14:29:41 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: *I've created the entries in cc_agents and cc_flows table.* *cc_agents table* +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | id | agentid | location | logstate | msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ *cc_flows table* +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | id | flowid | priority | skill | prependcid | max_wrapup_time | dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | dissuading_qsize_th | message_welcome | message_queue | message_dissuading | message_flow_id | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | 1 | support | 256 | support | NULL | 0 | 0 | 0 | 0 | 0 | | 112 at bp.erss.in | NULL | NULL | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ On Thu, 11 Apr 2024 at 14:27, Prathibha B wrote: > Getting the following error in call center module > > ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID > received) > Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: > ERROR:call_center:w_handle_call: failed to set new destination for call > > Call center code in opensips.cfg: > > if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { > if (!cc_handle_call("support")) { > send_reply(403,"Cannot handle call"); > exit; > } > } > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 09:00:15 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 14:30:15 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: bp.erss.in - asterisk bp.erss.in:1443 - opensips On Thu, 11 Apr 2024 at 14:29, Prathibha B wrote: > *I've created the entries in cc_agents and cc_flows table.* > > *cc_agents table* > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | id | agentid | location | logstate | msrp_location | > msrp_max_sessions | skills | wrapup_end_time | wrapup_time | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > > *cc_flows table* > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | id | flowid | priority | skill | prependcid | max_wrapup_time | > dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | > dissuading_qsize_th | message_welcome | message_queue | message_dissuading > | message_flow_id | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | 1 | support | 256 | support | NULL | 0 | > 0 | 0 | 0 | > 0 | | 112 at bp.erss.in | NULL | NULL > | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > > On Thu, 11 Apr 2024 at 14:27, Prathibha B > wrote: > >> Getting the following error in call center module >> >> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >> received) >> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >> ERROR:call_center:w_handle_call: failed to set new destination for call >> >> Call center code in opensips.cfg: >> >> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >> if (!cc_handle_call("support")) { >> send_reply(403,"Cannot handle call"); >> exit; >> } >> } >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Thu Apr 11 09:36:10 2024 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 11 Apr 2024 11:36:10 +0200 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Hi, Something wrong about that part for me : ($(tU) == "112") Should be : ($tU == "112") Regards De : Users au nom de Prathibha B Répondre à : OpenSIPS users mailling list Date : jeudi 11 avril 2024 à 11:03 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Call Center error bp.erss.in - asterisk bp.erss.in:1443 - opensips On Thu, 11 Apr 2024 at 14:29, Prathibha B wrote: I've created the entries in cc_agents and cc_flows table. cc_agents table +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | id | agentid | location | logstate | msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ cc_flows table +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | id | flowid | priority | skill | prependcid | max_wrapup_time | dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | dissuading_qsize_th | message_welcome | message_queue | message_dissuading | message_flow_id | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | 1 | support | 256 | support | NULL | 0 | 0 | 0 | 0 | 0 | | 112 at bp.erss.in | NULL | NULL | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ On Thu, 11 Apr 2024 at 14:27, Prathibha B wrote: Getting the following error in call center module ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: ERROR:call_center:w_handle_call: failed to set new destination for call Call center code in opensips.cfg: if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { if (!cc_handle_call("support")) { send_reply(403,"Cannot handle call"); exit; } } -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 09:39:08 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 15:09:08 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Errors in syslog: ERROR:core:parse_uri: bad uri, state 0 parsed: <112@> (4) / <112 at bp.erss.in;cc_pos=0> (23) ERROR:tm:uri2proxy: bad_uri: 112 at bp.erss.in;cc_pos=0 ERROR:b2b_entities:_client_new: while sending request with t_request ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b client instance ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) ERROR:call_center:w_handle_call: failed to set new destination for call ERROR:tm:_reply_light: failed to generate 403 reply when a final 408 was sent out ERROR:signaling:sig_send_reply_mod: failed to send reply with tm module On Thu, 11 Apr 2024 at 14:30, Prathibha B wrote: > bp.erss.in - asterisk > bp.erss.in:1443 - opensips > > On Thu, 11 Apr 2024 at 14:29, Prathibha B > wrote: > >> *I've created the entries in cc_agents and cc_flows table.* >> >> *cc_agents table* >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | id | agentid | location | logstate | msrp_location | >> msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> >> *cc_flows table* >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | id | flowid | priority | skill | prependcid | max_wrapup_time | >> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >> | message_flow_id | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | 1 | support | 256 | support | NULL | 0 | >> 0 | 0 | 0 | >> 0 | | 112 at bp.erss.in | NULL | NULL >> | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> >> On Thu, 11 Apr 2024 at 14:27, Prathibha B >> wrote: >> >>> Getting the following error in call center module >>> >>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >>> received) >>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>> ERROR:call_center:w_handle_call: failed to set new destination for call >>> >>> Call center code in opensips.cfg: >>> >>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>> if (!cc_handle_call("support")) { >>> send_reply(403,"Cannot handle call"); >>> exit; >>> } >>> } >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 09:40:32 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 15:10:32 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Cahnged it to $tU, Still getting error. On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent wrote: > Hi, > > > > Something wrong about that part for me : ($(tU) == "112") > > Should be : ($tU == "112") > > > > Regards > > > > *De : *Users au nom de Prathibha B < > prathibhab.tvm at gmail.com> > *Répondre à : *OpenSIPS users mailling list > *Date : *jeudi 11 avril 2024 à 11:03 > *À : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] Call Center error > > > > bp.erss.in - asterisk > > bp.erss.in:1443 - opensips > > > > On Thu, 11 Apr 2024 at 14:29, Prathibha B > wrote: > > *I've created the entries in cc_agents and cc_flows table.* > > > > *cc_agents table* > > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | id | agentid | location | logstate | msrp_location | > msrp_max_sessions | skills | wrapup_end_time | wrapup_time | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > > > > *cc_flows table* > > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | id | flowid | priority | skill | prependcid | max_wrapup_time | > dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | > dissuading_qsize_th | message_welcome | message_queue | message_dissuading > | message_flow_id | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | 1 | support | 256 | support | NULL | 0 | > 0 | 0 | 0 | > 0 | | 112 at bp.erss.in | NULL | NULL > | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > > > > On Thu, 11 Apr 2024 at 14:27, Prathibha B > wrote: > > Getting the following error in call center module > > > > ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID > received) > Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: > ERROR:call_center:w_handle_call: failed to set new destination for call > > > > Call center code in opensips.cfg: > > > > if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { > if (!cc_handle_call("support")) { > send_reply(403,"Cannot handle call"); > exit; > } > > } > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 10:03:09 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 15:33:09 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: when I run opensips-cli -x mi cc_list_agents { "Agents": [ { "id": "101002", "Ref": 0, "Loged in": "NO" }, { "id": "101001", "Ref": 0, "Loged in": "NO" } ] } I've logged in 101001 and 101002 in the browser. But the Loged in status is No for both users. On Thu, 11 Apr 2024 at 15:10, Prathibha B wrote: > Cahnged it to $tU, Still getting error. > > On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent > wrote: > >> Hi, >> >> >> >> Something wrong about that part for me : ($(tU) == "112") >> >> Should be : ($tU == "112") >> >> >> >> Regards >> >> >> >> *De : *Users au nom de Prathibha B < >> prathibhab.tvm at gmail.com> >> *Répondre à : *OpenSIPS users mailling list >> *Date : *jeudi 11 avril 2024 à 11:03 >> *À : *OpenSIPS users mailling list >> *Objet : *Re: [OpenSIPS-Users] Call Center error >> >> >> >> bp.erss.in - asterisk >> >> bp.erss.in:1443 - opensips >> >> >> >> On Thu, 11 Apr 2024 at 14:29, Prathibha B >> wrote: >> >> *I've created the entries in cc_agents and cc_flows table.* >> >> >> >> *cc_agents table* >> >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | id | agentid | location | logstate | msrp_location | >> msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> >> >> >> *cc_flows table* >> >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | id | flowid | priority | skill | prependcid | max_wrapup_time | >> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >> | message_flow_id | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | 1 | support | 256 | support | NULL | 0 | >> 0 | 0 | 0 | >> 0 | | 112 at bp.erss.in | NULL | NULL >> | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> >> >> >> On Thu, 11 Apr 2024 at 14:27, Prathibha B >> wrote: >> >> Getting the following error in call center module >> >> >> >> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >> received) >> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >> ERROR:call_center:w_handle_call: failed to set new destination for call >> >> >> >> Call center code in opensips.cfg: >> >> >> >> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >> if (!cc_handle_call("support")) { >> send_reply(403,"Cannot handle call"); >> exit; >> } >> >> } >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> _______________________________________________ Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Thu Apr 11 10:46:58 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Thu, 11 Apr 2024 16:16:58 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. Now getting the following errors in syslog: ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change URI Contact ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change URI Contact ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was sent out ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, [B2B.394.162.1712831164.913185781] On Thu, 11 Apr 2024 at 15:33, Prathibha B wrote: > when I run opensips-cli -x mi cc_list_agents > { > "Agents": [ > { > "id": "101002", > "Ref": 0, > "Loged in": "NO" > }, > { > "id": "101001", > "Ref": 0, > "Loged in": "NO" > } > ] > } > > I've logged in 101001 and 101002 in the browser. But the Loged in status > is No for both users. > > On Thu, 11 Apr 2024 at 15:10, Prathibha B > wrote: > >> Cahnged it to $tU, Still getting error. >> >> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >> wrote: >> >>> Hi, >>> >>> >>> >>> Something wrong about that part for me : ($(tU) == "112") >>> >>> Should be : ($tU == "112") >>> >>> >>> >>> Regards >>> >>> >>> >>> *De : *Users au nom de Prathibha B < >>> prathibhab.tvm at gmail.com> >>> *Répondre à : *OpenSIPS users mailling list >>> *Date : *jeudi 11 avril 2024 à 11:03 >>> *À : *OpenSIPS users mailling list >>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>> >>> >>> >>> bp.erss.in - asterisk >>> >>> bp.erss.in:1443 - opensips >>> >>> >>> >>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>> wrote: >>> >>> *I've created the entries in cc_agents and cc_flows table.* >>> >>> >>> >>> *cc_agents table* >>> >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> | id | agentid | location | logstate | msrp_location | >>> msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>> | 4 | support | 0 | 0 | >>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>> | 4 | support | 0 | 0 | >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> >>> >>> >>> *cc_flows table* >>> >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>> | message_flow_id | >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> | 1 | support | 256 | support | NULL | 0 | >>> 0 | 0 | 0 | >>> 0 | | 112 at bp.erss.in | NULL | NULL >>> | >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> >>> >>> >>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>> wrote: >>> >>> Getting the following error in call center module >>> >>> >>> >>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >>> received) >>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>> ERROR:call_center:w_handle_call: failed to set new destination for call >>> >>> >>> >>> Call center code in opensips.cfg: >>> >>> >>> >>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>> if (!cc_handle_call("support")) { >>> send_reply(403,"Cannot handle call"); >>> exit; >>> } >>> >>> } >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> _______________________________________________ Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 11 12:23:57 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 11 Apr 2024 14:23:57 +0200 Subject: [OpenSIPS-Users] another question on sip format's. Message-ID: Can param's be added anywhere in a user part. e.g. sip:0123456789;gw=case106 at 1.2.3.4 //this is for sure valid. sip:gw=case107;0123456789 at 1.2.3.4 sip:gw=case108;0123456789;gw=case108 at 1.2.3.4 the fist one is for sure valid, but what about the other two ? BR, -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 11 14:24:08 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 11 Apr 2024 17:24:08 +0300 Subject: [OpenSIPS-Users] another question on sip format's. In-Reply-To: References: Message-ID: <95dfc7cb-a30d-4449-8122-83453b3035d6@opensips.org> According to the grammar, the URI params are ONLY after the domain part (see https://www.ietf.org/rfc/rfc3261.html page 222) SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] SIPS-URI = "sips:" [ userinfo ] hostport uri-parameters [ headers ] Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11.04.2024 15:23, Johan De Clercq wrote: > Can param's be added anywhere in a user part. > > e.g. > sip:0123456789;gw=case106 at 1.2.3.4 //this is for sure valid. > sip:gw=case107;0123456789 at 1.2.3.4 > sip:gw=case108;0123456789;gw=case108 at 1.2.3.4 > > the fist one is for sure valid, but what about the other two ? > > BR, > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Apr 11 16:07:23 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 11 Apr 2024 16:07:23 +0000 Subject: [OpenSIPS-Users] another question on sip format's. In-Reply-To: <95dfc7cb-a30d-4449-8122-83453b3035d6@opensips.org> References: <95dfc7cb-a30d-4449-8122-83453b3035d6@opensips.org> Message-ID: As far as I can tell though all the examples are valid for the userinfo portion of the URI. SIP-URI = sip: [ userinfo ] hostport uri-parameters [ headers ] SIPS-URI = sips: [ userinfo ] hostport uri-parameters [ headers ] userinfo = ( user / telephone-subscriber ) [ ":" password ] "@" user = 1*( unreserved / escaped / user-unreserved ) user-unreserved = "&" / "=" / "+" / "$" / "," / ";" / "?" / "/" password = *( unreserved / escaped / "&" / "=" / "+" / "$" / "," ) As far as I know, user parameters of this type aren’t explicitly used or defined by RFC 3261 for SIP URIs; they really come from tel URIs [1]. But their use in SIP is widespread and defined in various other RFCs as well. [1] - https://datatracker.ietf.org/doc/html/rfc3966 Ben Newlin From: Users on behalf of Bogdan-Andrei Iancu Date: Thursday, April 11, 2024 at 10:26 AM To: OpenSIPS users mailling list , Johan De Clercq Subject: Re: [OpenSIPS-Users] another question on sip format's. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ According to the grammar, the URI params are ONLY after the domain part (see https://www.ietf.org/rfc/rfc3261.html page 222) SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] SIPS-URI = "sips:" [ userinfo ] hostport uri-parameters [ headers ] Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11.04.2024 15:23, Johan De Clercq wrote: Can param's be added anywhere in a user part. e.g. sip:0123456789;gw=case106 at 1.2.3.4 //this is for sure valid. sip:gw=case107;0123456789 at 1.2.3.4 sip:gw=case108;0123456789;gw=case108 at 1.2.3.4 the fist one is for sure valid, but what about the other two ? BR, _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Thu Apr 11 17:07:48 2024 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Thu, 11 Apr 2024 13:07:48 -0400 Subject: [OpenSIPS-Users] $Ri Variable Message-ID: Hello! I am trying to get the IP of the interface the request was received on, but I think I am doing something wrong. When I looked at the Core Variables, I found $Ri. So, I added $Ri to the cfg file, but now I am getting: unknown script var $Ri, maybe a 'loadmodule' statement is missing? Which module am I missing? Or, am I even referencing the correct variable? Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Fri Apr 12 02:41:31 2024 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Thu, 11 Apr 2024 22:41:31 -0400 Subject: [OpenSIPS-Users] Load Balancer Probing Question Message-ID: Hi. I have an interesting issue. We have two OpenSIPS servers with load balancer (with two different group IDs in the lb table) and we have probing set correctly and we are using the event, E_LOAD_BALANCER_STATUS, to capture changes to servers that were probed. But we noticed that we have the same server URI listed in the lb table, but with two different group IDs, if one of the OpenSIPS servers probes that URI and it does not return, then lb disables both groups. I'd expect it to only disable one group. My question is how can we tell the LB module to disable the IP, but also look for the groupID. For example, I have a printout of lb_list below. "uri": "sip:1.2.3.4:5060", "id": 27, "group": 12, "enabled": "no", "auto-reenable": "on", "Resources": [ { "name": "vz12", "max": 600, "load": 0 } ], "attrs": "0" AND { "uri": "sip:1.2.3.4:5060", "id": 29, "group": 13, "enabled": "no", "auto-reenable": "on", "Resources": [ { "name": "vz13", "max": 600, "load": 0 } ], "attrs": "0" }, Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Fri Apr 12 10:34:56 2024 From: ulus_egemen at hotmail.com (egemen ulus) Date: Fri, 12 Apr 2024 10:34:56 +0000 Subject: [OpenSIPS-Users] Callee details not visible in dialog detail In-Reply-To: References: Message-ID: Sorry, you're right. Please ignore this. Regards Get Outlook for Android ________________________________ From: Brett Nemeroff Sent: Friday, March 29, 2024 6:16:02 PM To: OpenSIPS users mailling list Cc: egemen ulus Subject: Re: [OpenSIPS-Users] Callee details not visible in dialog detail While I agree with Bogdan about asking on the right list this particular one is likely to be resolved the same way. You gotta be careful because there are enough differences where that isn’t always the case. I believe dlg_manage is Kamailio so you are definitely in the wrong spot! Dialog variables can be used to store just about anything into the dialog works similarly on both. Best of luck! https://www.kamailio.org/docs/modules/devel/modules/dialog.html#idm1681 https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp6006096 On Thu, Mar 28, 2024 at 3:36 PM Bogdan-Andrei Iancu > wrote: Hi, If using Kamailio, you should ask this on the kamilio mailing list, right ? This mailing list is for OpenSIPS. Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 28.03.2024 16:35, egemen ulus wrote: Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. Any way to have asterisk IP in dialog details? It shows only this: { h_entry: 483 h_id: 8644 ref: 1 call-id: v489LdfOFD5mnlHupQHOvw.. from_uri: sip:1515 at 12.12.12.12;transport=UDP to_uri: sip:2222 at 12.12.12.12 state: 5 start_ts: 0 init_ts: xxxx end_ts: xxxx duration: xxxx timeout: 0 lifetime: 3600 dflags: 512 sflags: 0 iflags: 0 caller: { tag: d0283007 contact: sip:1515 at 1.1.1.1:64966;transport=UDP cseq: 1 route_set: socket: udp:172.22.10.10:5060 } callee: { tag: contact: cseq: route_set: socket: } profiles: { } variables: { } } Regards Get Outlook for Android _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmed.rehan at gmail.com Mon Apr 15 19:48:36 2024 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Tue, 16 Apr 2024 00:48:36 +0500 Subject: [OpenSIPS-Users] AUTH_JWT Message-ID: I m trying to use jwt_script_authorize in AUTH JWT module . can i have a clue the pub key to pass in the this function to validate JWT is in PEM format or any other format ? -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From slackway2me at gmail.com Tue Apr 16 06:49:49 2024 From: slackway2me at gmail.com (Alexey) Date: Tue, 16 Apr 2024 11:49:49 +0500 Subject: [OpenSIPS-Users] dropped calls In-Reply-To: References: Message-ID: Catch CANCEL requests which are sent by clients, according to 9.1 section of the RFC [1]. [1] https://datatracker.ietf.org/doc/html/rfc3261#section-9.1 -- best regards, Alexey https://alexeyka.zantsev.com/ From bogdan at opensips.org Tue Apr 16 07:25:26 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 16 Apr 2024 10:25:26 +0300 Subject: [OpenSIPS-Users] $Ri Variable In-Reply-To: References: Message-ID: Hi, Check this https://opensips.org/Documentation/Script-CoreVar-3-4#socket_in Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11.04.2024 20:07, Alexander Perkins wrote: > Hello!  I am trying to get the IP of the interface the request was > received on, but I think I am doing something wrong.  When I looked at > the Core Variables, I found $Ri.  So, I added $Ri to the cfg file, but > now I am getting: > > unknown script var $Ri, maybe a 'loadmodule' statement is missing? > > Which module am I missing?  Or, am I even referencing the correct > variable? > > Thank you, > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Apr 16 07:28:42 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 16 Apr 2024 10:28:42 +0300 Subject: [OpenSIPS-Users] Load Balancer Probing Question In-Reply-To: References: Message-ID: <3982635b-42e7-4a81-bdc6-5a3ed47ccfc0@opensips.org> Hi, What OpenSIPS version you have? And as I understand, as configuration, you do permanent probing to the destinations and the disabling happens because of this probing ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 12.04.2024 05:41, Alexander Perkins wrote: > Hi.  I have an interesting issue.  We have two OpenSIPS servers with > load balancer (with two different group IDs in the lb table) and we > have probing set correctly and we are using the > event, E_LOAD_BALANCER_STATUS, to capture changes to servers that were > probed.  But we noticed that we have the same server URI listed in the > lb table, but with two different group IDs, if one of the OpenSIPS > servers probes that URI and it does not return, then lb disables both > groups.  I'd expect it to only disable one group. > > My question is how can we tell the LB module to disable the IP, but > also look for the groupID.  For example, I have a printout of lb_list > below. > > "uri": "sip:1.2.3.4:5060 ", "id": 27, "group": > 12,"enabled": "no", "auto-reenable": "on", "Resources": [ { "name": > "vz12", "max": 600, "load": 0 } ], "attrs": "0" > > AND > { "uri": "sip:1.2.3.4:5060 ", "id": 29, "group": > 13,"enabled": "no", "auto-reenable": "on", "Resources": [ { "name": > "vz13", "max": 600, "load": 0 } ], "attrs": "0" }, > > Thank you, > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Apr 16 07:45:08 2024 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 16 Apr 2024 08:45:08 +0100 Subject: [OpenSIPS-Users] Load Balancer Probing Question In-Reply-To: <3982635b-42e7-4a81-bdc6-5a3ed47ccfc0@opensips.org> References: <3982635b-42e7-4a81-bdc6-5a3ed47ccfc0@opensips.org> Message-ID: If the backend servers are both the same instance then this seems to be the correct behaviour? I believe the probing is supposed to be a simple SIP response healthcheck which applies to the destination globally (i.e. 1.2.3.4 is offline), the groups are just a way of splitting up resources logically for load balancing purposes. On Tue, 16 Apr 2024 at 08:31, Bogdan-Andrei Iancu wrote: > Hi, > > What OpenSIPS version you have? And as I understand, as configuration, you > do permanent probing to the destinations and the disabling happens because > of this probing ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 12.04.2024 05:41, Alexander Perkins wrote: > > Hi. I have an interesting issue. We have two OpenSIPS servers with load > balancer (with two different group IDs in the lb table) and we have probing > set correctly and we are using the event, E_LOAD_BALANCER_STATUS, to > capture changes to servers that were probed. But we noticed that we have > the same server URI listed in the lb table, but with two different group > IDs, if one of the OpenSIPS servers probes that URI and it does not return, > then lb disables both groups. I'd expect it to only disable one group. > > My question is how can we tell the LB module to disable the IP, but also > look for the groupID. For example, I have a printout of lb_list below. > > "uri": "sip:1.2.3.4:5060", "id": 27, "group": 12, "enabled": "no", > "auto-reenable": "on", "Resources": [ { "name": "vz12", "max": 600, "load": > 0 } ], "attrs": "0" > > AND > { "uri": "sip:1.2.3.4:5060", "id": 29, "group": 13, "enabled": "no", > "auto-reenable": "on", "Resources": [ { "name": "vz13", "max": 600, "load": > 0 } ], "attrs": "0" }, > > Thank you, > Alex > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *0333 332 0000  |  x-on.co.uk   | **  |   **Practice Index Reviews * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of X-on Health Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 16 10:58:28 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 16 Apr 2024 16:28:28 +0530 Subject: [OpenSIPS-Users] send failed Message-ID: You tried t make a video call. send failed(477/TM). I am getting this error frequently. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Tue Apr 16 14:09:16 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Tue, 16 Apr 2024 19:39:16 +0530 Subject: [OpenSIPS-Users] Launch and async Message-ID: Hello, I want to use Launch and async function for function sleep() as I need to wait for some time to do some operations but from the route which I use in above function I am not able to use any variable for my operation and from documentation it seems only able to use those variable which sleep() function use so is there any alternative solution to get variable??? or any other function on which I can wait for a specific time without blocking opensips?? Regards, *Parthesh Bhavsar | Software Engineer | VOIP* -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Tue Apr 16 14:25:56 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Tue, 16 Apr 2024 19:55:56 +0530 Subject: [OpenSIPS-Users] Launch and async In-Reply-To: References: Message-ID: I'm facing a challenge in my OpenSIPS implementation where I need to perform certain operations after receiving a 200 OK response, but only after waiting for 10 seconds. Currently, I'm considering using the sleep() function, but I'm concerned about its impact on performance. Is there a recommended approach to achieve this functionality without compromising OpenSIPS performance? I'd appreciate any advice or alternative solutions you can offer. Regards, *Parthesh Bhavsar | Software Engineer | VOIP* On Tue, Apr 16, 2024 at 7:39 PM Parthesh Bhavsar < parthesh.bhavsar at ecosmob.com> wrote: > Hello, > I want to use Launch and async function for function sleep() as I need to > wait for some time to do some operations but from the route which I use in > above function I am not able to use any variable for my operation and from > documentation it seems only able to use those variable which sleep() > function use so is there any alternative solution to get variable??? or any > other function on which I can wait for a specific time without blocking > opensips?? > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 18 07:15:28 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 18 Apr 2024 10:15:28 +0300 Subject: [OpenSIPS-Users] Launch and async In-Reply-To: References: Message-ID: Hi, The only way to do a non-blocking waiting is via the async(sleep()) combination, but the async() statement works only in main request route (in route{}) and not for reply route :( . Doing a blocking sleep() it is very very dangerous in terms of performance, as you will block the opensips processes with these sleep()'s. Anyhow, from SIP perspective it is not wise to delay a reply, as this will trigger retransmissions from the UAC side....so, not sure if what you try to do is actually something right. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 16.04.2024 17:25, Parthesh Bhavsar via Users wrote: > I'm facing a challenge in my OpenSIPS implementation where I need to > perform certain operations after receiving a 200 OK response, but only > after waiting for 10 seconds. Currently, I'm considering using the > sleep() function, but I'm concerned about its impact on performance. > > Is there a recommended approach to achieve this functionality without > compromising OpenSIPS performance? I'd appreciate any advice or > alternative solutions you can offer. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > On Tue, Apr 16, 2024 at 7:39 PM Parthesh Bhavsar > wrote: > > Hello, > I want to use Launch and async function for function sleep() as I > need to wait for some time to do some operations but from the > route which I use in above function I am not able to use any > variable for my operation and from documentation it seems > only able to use those variable which sleep() function use so is > there any alternative solution to get variable??? or any other > function on which I can wait for a specific time without blocking > opensips?? > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > *https://www.ecosmob.com/opensips-summit/ > > * > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our > website, any views or opinions presented in this email are solely > those of the originator and do not necessarily represent those of the > Company or its sister concerns. Any liability (in negligence, contract > or otherwise) arising from any third party taking any action, or > refraining from taking any action on the basis of any of the > information contained in this email is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for > the use of the addressee(s) and contains information that is > PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, > distribution, or copying of this communication is prohibited. Please > inform originator if you have received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of > viruses, trojans, similar or new contaminants/malware, interceptions > or interference, and may not be compatible with your systems. You > shall carry out virus/malware scanning on your own before opening any > attachment to this e-mail. The sender of this e-mail and Company > including its sister concerns shall not be liable for any damage that > may incur to you as a result of viruses, incompleteness of this > message, a delay in receipt of this message or any other computer > problems. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 18 07:25:43 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 18 Apr 2024 10:25:43 +0300 Subject: [OpenSIPS-Users] Load Balancer Probing Question In-Reply-To: References: <3982635b-42e7-4a81-bdc6-5a3ed47ccfc0@opensips.org> Message-ID: Not really. The LB module uses internal unique ids for all the LB destinations it manages. So the probing replies will search back the LB destination based on this ID -> no chance to mismatch. Regards. Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 16.04.2024 10:45, Callum Guy via Users wrote: > If the backend servers are both the same instance then this seems to > be the correct behaviour? > > I believe the probing is supposed to be a simple SIP response > healthcheck which applies to the destination globally (i.e. 1.2.3.4 is > offline), the groups are just a way of splitting up resources > logically for load balancing purposes. > > On Tue, 16 Apr 2024 at 08:31, Bogdan-Andrei Iancu > wrote: > > Hi, > > What OpenSIPS version you have? And as I understand, as > configuration, you do permanent probing to the destinations and > the disabling happens because of this probing ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 12.04.2024 05:41, Alexander Perkins wrote: >> Hi.  I have an interesting issue.  We have two OpenSIPS servers >> with load balancer (with two different group IDs in the lb table) >> and we have probing set correctly and we are using the >> event, E_LOAD_BALANCER_STATUS, to capture changes to servers that >> were probed.  But we noticed that we have the same server URI >> listed in the lb table, but with two different group IDs, if one >> of the OpenSIPS servers probes that URI and it does not return, >> then lb disables both groups.  I'd expect it to only disable one >> group. >> >> My question is how can we tell the LB module to disable the IP, >> but also look for the groupID.  For example, I have a printout of >> lb_list below. >> >> "uri": "sip:1.2.3.4:5060 ", "id": 27, >> "group": 12,"enabled": "no", "auto-reenable": "on", "Resources": >> [ { "name": "vz12", "max": 600, "load": 0 } ], "attrs": "0" >> >> AND >> { "uri": "sip:1.2.3.4:5060 ", "id": 29, >> "group": 13,"enabled": "no", "auto-reenable": "on", "Resources": >> [ { "name": "vz13", "max": 600, "load": 0 } ], "attrs": "0" }, >> >> Thank you, >> Alex >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > *^0333 332 0000  | x-on.co.uk   | **^  | > **^Practice Index Reviews * > > *Our new office address: 22 Riduna Park, Melton IP12 1QT.* > > X-on is a trading name of X-on Health Ltd a limited company registered > in England and Wales. > Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, > Hampshire, England RG25 2AD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must > not use, disclose, disseminate, distribute, copy, print or reply to > this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 18 08:32:39 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 18 Apr 2024 10:32:39 +0200 Subject: [OpenSIPS-Users] question on core statistics. Message-ID: Guys, I have an opensips instance running with 24 worker children. The worker load is very low. UDP queues are on 50 megs. when i query via the OS cat /proc/net/udp sl local_address rem_address st tx_queue rx_queue tr tm->when retrnsmt uid timeout inode ref pointer drops 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413684019 2 ffff880074820bc0 0 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413766438 2 ffff880465e4a440 0 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 412035865 2 ffff8803e5a56b80 0 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26790 2 ffff88046c054840 0 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26787 2 ffff88046c054bc0 0 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26791 2 ffff88046c0544c0 0 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15506 2 ffff88046dce5040 0 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 22811 2 ffff880465e4ab40 0 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20572 2 ffff88086d020800 0 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20571 2 ffff88086d020b80 0 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 00000000 100 0 17515 2 ffff8800368ac780 0 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15505 2 ffff8800368acb00 0 => no drops what worries me is that there are drop requests and they go up when I query via the mi interface opensipsctl fifo get_statistics drop_requests core:drop_requests:: 198107 opensipsctl fifo get_statistics drop_requests core:drop_requests:: 199157 opensipsctl_reg fifo get_statistics drop_requests core:drop_requests:: 204116 I don't see any memory issue, also the processload is low. so 3 questions: - what exactly is drop_request. - do I need to worry about this - how can I make them go lower. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 18 09:07:15 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 18 Apr 2024 11:07:15 +0200 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: would it make sense to recompile with other flags ? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. Op do 18 apr 2024 om 10:32 schreef Johan De Clercq : > > Guys, > > I have an opensips instance running with 24 worker children. > The worker load is very low. > UDP queues are on 50 megs. > > when i query via the OS > cat /proc/net/udp > sl local_address rem_address st tx_queue rx_queue tr tm->when > retrnsmt uid timeout inode ref pointer drops > 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413684019 2 ffff880074820bc0 0 > 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413766438 2 ffff880465e4a440 0 > 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 412035865 2 ffff8803e5a56b80 0 > 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26790 2 ffff88046c054840 0 > 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26787 2 ffff88046c054bc0 0 > 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26791 2 ffff88046c0544c0 0 > 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15506 2 ffff88046dce5040 0 > 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 22811 2 ffff880465e4ab40 0 > 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20572 2 ffff88086d020800 0 > 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20571 2 ffff88086d020b80 0 > 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 100 0 17515 2 ffff8800368ac780 0 > 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15505 2 ffff8800368acb00 0 > > => no drops > > what worries me is that there are drop requests and they go up when I > query via the mi interface > opensipsctl fifo get_statistics drop_requests > core:drop_requests:: 198107 > opensipsctl fifo get_statistics drop_requests > core:drop_requests:: 199157 > opensipsctl_reg fifo get_statistics drop_requests > core:drop_requests:: 204116 > > I don't see any memory issue, also the processload is low. > > > so 3 questions: > - what exactly is drop_request. > - do I need to worry about this > - how can I make them go lower. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 18 09:25:55 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 18 Apr 2024 11:25:55 +0200 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: would it make sense to recompile with other flags ? And how do I set them (I don't find these of menuconfig's compile options)? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. Op do 18 apr 2024 om 11:07 schreef Johan De Clercq : > would it make sense to recompile with other flags ? > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > Can somebody explain also what both flags mean. > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll, sigio_rt, select. > > Op do 18 apr 2024 om 10:32 schreef Johan De Clercq : > >> >> Guys, >> >> I have an opensips instance running with 24 worker children. >> The worker load is very low. >> UDP queues are on 50 megs. >> >> when i query via the OS >> cat /proc/net/udp >> sl local_address rem_address st tx_queue rx_queue tr tm->when >> retrnsmt uid timeout inode ref pointer drops >> 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 413684019 2 ffff880074820bc0 0 >> 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 413766438 2 ffff880465e4a440 0 >> 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 412035865 2 ffff8803e5a56b80 0 >> 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 26790 2 ffff88046c054840 0 >> 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 26787 2 ffff88046c054bc0 0 >> 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 26791 2 ffff88046c0544c0 0 >> 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 15506 2 ffff88046dce5040 0 >> 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 22811 2 ffff880465e4ab40 0 >> 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 20572 2 ffff88086d020800 0 >> 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 20571 2 ffff88086d020b80 0 >> 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 100 0 17515 2 ffff8800368ac780 0 >> 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 >> 00000000 0 0 15505 2 ffff8800368acb00 0 >> >> => no drops >> >> what worries me is that there are drop requests and they go up when I >> query via the mi interface >> opensipsctl fifo get_statistics drop_requests >> core:drop_requests:: 198107 >> opensipsctl fifo get_statistics drop_requests >> core:drop_requests:: 199157 >> opensipsctl_reg fifo get_statistics drop_requests >> core:drop_requests:: 204116 >> >> I don't see any memory issue, also the processload is low. >> >> >> so 3 questions: >> - what exactly is drop_request. >> - do I need to worry about this >> - how can I make them go lower. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Apr 18 13:14:59 2024 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 18 Apr 2024 16:14:59 +0300 Subject: [OpenSIPS-Users] [Minor Release] OpenSIPS 3.4.5 and 3.2.18 Minor Releases Message-ID: <1fa11025-9576-1110-e1d3-487efc42b219@opensips.org> Hi, everyone! A new round of stable minor releases is now out: *3.4.5 *and *3.2.18*, which include all fixes done in the past two months. Do make sure to schedule an update as soon as possible! Full changelogs: https://opensips.org/pub/opensips/3.4.5/ChangeLog https://opensips.org/pub/opensips/3.2.18/ChangeLog Please enjoy! OpenSIPS Team -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Apr 18 13:58:53 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 18 Apr 2024 13:58:53 +0000 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: Are you calling drop() anywhere in your script? https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 Ben Newlin From: Users on behalf of Johan De Clercq Date: Thursday, April 18, 2024 at 5:27 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] question on core statistics. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ would it make sense to recompile with other flags ? And how do I set them (I don't find these of menuconfig's compile options)? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >: would it make sense to recompile with other flags ? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >: Guys, I have an opensips instance running with 24 worker children. The worker load is very low. UDP queues are on 50 megs. when i query via the OS cat /proc/net/udp sl local_address rem_address st tx_queue rx_queue tr tm->when retrnsmt uid timeout inode ref pointer drops 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413684019 2 ffff880074820bc0 0 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413766438 2 ffff880465e4a440 0 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 412035865 2 ffff8803e5a56b80 0 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26790 2 ffff88046c054840 0 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26787 2 ffff88046c054bc0 0 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26791 2 ffff88046c0544c0 0 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15506 2 ffff88046dce5040 0 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 22811 2 ffff880465e4ab40 0 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20572 2 ffff88086d020800 0 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20571 2 ffff88086d020b80 0 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 00000000 100 0 17515 2 ffff8800368ac780 0 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15505 2 ffff8800368acb00 0 => no drops what worries me is that there are drop requests and they go up when I query via the mi interface opensipsctl fifo get_statistics drop_requests core:drop_requests:: 198107 opensipsctl fifo get_statistics drop_requests core:drop_requests:: 199157 opensipsctl_reg fifo get_statistics drop_requests core:drop_requests:: 204116 I don't see any memory issue, also the processload is low. so 3 questions: - what exactly is drop_request. - do I need to worry about this - how can I make them go lower. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wadii at evenmedia.fr Thu Apr 18 14:09:07 2024 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Thu, 18 Apr 2024 14:09:07 +0000 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: Calling exit() during the initial request and right before creating the dialog also increments the drop_requests statistic De : Users De la part de Ben Newlin Envoyé : jeudi 18 avril 2024 15:59 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] question on core statistics. Are you calling drop() anywhere in your script? https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 Ben Newlin From: Users > on behalf of Johan De Clercq > Date: Thursday, April 18, 2024 at 5:27 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] question on core statistics. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ would it make sense to recompile with other flags ? And how do I set them (I don't find these of menuconfig's compile options)? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >: would it make sense to recompile with other flags ? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >: Guys, I have an opensips instance running with 24 worker children. The worker load is very low. UDP queues are on 50 megs. when i query via the OS cat /proc/net/udp sl local_address rem_address st tx_queue rx_queue tr tm->when retrnsmt uid timeout inode ref pointer drops 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413684019 2 ffff880074820bc0 0 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413766438 2 ffff880465e4a440 0 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 412035865 2 ffff8803e5a56b80 0 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26790 2 ffff88046c054840 0 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26787 2 ffff88046c054bc0 0 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26791 2 ffff88046c0544c0 0 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15506 2 ffff88046dce5040 0 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 22811 2 ffff880465e4ab40 0 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20572 2 ffff88086d020800 0 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20571 2 ffff88086d020b80 0 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 00000000 100 0 17515 2 ffff8800368ac780 0 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15505 2 ffff8800368acb00 0 => no drops what worries me is that there are drop requests and they go up when I query via the mi interface opensipsctl fifo get_statistics drop_requests core:drop_requests:: 198107 opensipsctl fifo get_statistics drop_requests core:drop_requests:: 199157 opensipsctl_reg fifo get_statistics drop_requests core:drop_requests:: 204116 I don't see any memory issue, also the processload is low. so 3 questions: - what exactly is drop_request. - do I need to worry about this - how can I make them go lower. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Apr 4 14:10:28 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 4 Apr 2024 14:10:28 +0000 Subject: [OpenSIPS-Users] dashboard stats from opensips In-Reply-To: References: <24073738-3ed2-4996-ac49-ab1124b1e217@opensips.org> Message-ID: All of the dlg_vals should be available after calling load_dialog_ctx. We are using it that way in our system. Ben Newlin From: Users on behalf of nz deals Date: Thursday, April 4, 2024 at 9:56 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Much appreciated Ben, I was able to test $params so all is good with them. I was not able to access attributes which I have set after the create_dialog. create_dialog(); $dlg_val(caller) = $fu; $dlg_val(callee) = $ru; i also wanted to access $dlg_val(caller) and $dlg_val(callee) under event_route[E_DLG_STATE_CHANGED] On Thu, 4 Apr 2024 at 11:10, Ben Newlin > wrote: The parameters exposed by the E_DLG_STATE_CHANGED event are documented [1]. They are accessed using the $param notation [2]. You can then use load_dialog_ctx [3] and get all the other dialog information you need. [1] - https://opensips.org/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED. [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-4#param [3] - https://opensips.org/docs/modules/3.2.x/dialog.html#func_load_dialog_ctx Ben Newlin From: Users > on behalf of nz deals > Date: Wednesday, April 3, 2024 at 5:38 PM To: OpenSIPS users mailling list >, Bogdan-Andrei Iancu > Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Thank you, Brett. My thoughts have been on events and Redis ;) I'll also explore RabbitMQ, thanks for the suggestion. From what I gather, we can utilize E_DLG_STATE_CHANGED within the event route. Could you guide me on how to retrieve the dialogid/callid and its state? If I can access this information in the event_route[E_DLG_STATE_CHANGED], storing the value would be straightforward. Thanks On Thu, 4 Apr 2024 at 10:13, Brett Nemeroff > wrote: I'd recommend using the events and rabbitmq. You should be able to do just about anything with that. What cps are you processing? On Wed, Apr 3, 2024 at 3:46 PM nz deals > wrote: Thanks Ben, The issue with the scheduled task is that it introduces a delay. I'm exploring methods to enable real-time display. By streaming events directly from OpenSIPS, we could achieve live updates on the display. Thank you Regards, Jason On Thu, 4 Apr 2024 at 06:15, Ben Newlin > wrote: OpenSIPS will already track some very basic statistics like this for you using the Statistics module. https://opensips.org/docs/modules/3.4.x/statistics.html For example, the Dialog module exposes concurrent calls. We have a scheduled job that queries those stats via MI and pushes them into our external metrics system, allowing us to create dashboards from the data. Ben Newlin From: Users > on behalf of nz deals > Date: Wednesday, April 3, 2024 at 1:03 PM To: Bogdan-Andrei Iancu > Cc: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] dashboard stats from opensips EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi Bogdan, Yes, something along the lines of the OpenSIPs control panel, but I'm looking for very basic statistics, such as the details of currently active calls and a straightforward graph displaying concurrent calls. Thank you Regards, Jason On Thu, 4 Apr 2024 at 03:27, Bogdan-Andrei Iancu > wrote: Hi Jason, Have you checked the Dashboard in OpenSIPS Control Panel ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 03.04.2024 13:54, nz deals wrote: > Hi everyone, > > I'm seeking guidance on creating a dashboard. I'm considering saving > dialog events in Redis (straight from OpenSIPS), allowing my dashboard > to directly access the Redis cache. Do you think this is a wise > strategy, or do you have any alternative suggestions? Any expert's > suggestion will be highly appreciated. In fact if someone has any > example to check, raise dialog events like call, ringing, 183, > answered , cancel and bye etc... > > Thank you > > Regards, > Jason > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Fri Apr 5 01:29:55 2024 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 5 Apr 2024 01:29:55 +0000 Subject: [OpenSIPS-Users] external applications In-Reply-To: References: <5ecfbddd-d737-f2ca-b93d-a93c1886dff6@voipplus.net> <0831297d-6722-4bbc-ae09-b13fa76b649f@opensips.org> Message-ID: There is no generic answer to that question. It depends on your routing script and the way your system works. The code you posted before checking for INVITE could do it, but only if it is placed in the right location. You may want to review the different types of routes used in OpenSIPS scripting [1], and what kind of message or processing is being performed in each. That may help you determine where to place your checks. Routing of a SIP request always begins in the unnamed route block. [1] - https://www.opensips.org/Documentation/Script-Routes-3-4 Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 8:21 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ My question is not regarding exec command. How to capture the start of the call? I'm unable to get the start. Sent from Outlook for Android ________________________________ From: Users on behalf of Ben Newlin Sent: Thursday, April 4, 2024 9:22:28 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications Per the documentation [1], input variables to the script are provided as a separate parameter to the exec command. It does not do a direct shell execution of the entire first parameter. So it should be: exec(“script.sh”, “INVITE”); Also, for a relative command I’m not entirely sure where OpenSIPS would look. It could be the working directory or it could be the directory from which opensips was launched. Have you tried using an absolute path? Are you setting the wdir [2] parameter? [1] - https://opensips.org/docs/modules/3.4.x/exec.html#func_exec [2] - https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B > wrote: if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B > wrote: Yes. I am canceling the call prior to dialog creation. Sent from Outlook for Android ________________________________ From: Users > on behalf of Ben Newlin > Sent: Wednesday, April 3, 2024 7:03:33 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications If your script is cancelling the call then why wouldn’t you “capture it” in the same place? Send whatever you need to whatever external entity you are using directly. You don’t need a callback to trigger if you know you are taking the action. A created dialog being cancelled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users > on behalf of Prathibha B > Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I am capturing the dropped call after ringing in failure_route, but If I cancelled during the start of the call, how to capture it. On Tue, 2 Apr 2024 at 20:38, Ben Newlin > wrote: The start of the call would be when you call “create_dialog”. The dialog state for that is “UNCONFIRMED”. I’m not sure whether a dialog state change event is raised for creation. It may only be raised when the state changes after creation. But since you control the dialog creation, you can just take whatever action you desire at that time. Ben Newlin From: Users > on behalf of Prathibha B > Date: Tuesday, April 2, 2024 at 8:05 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B > wrote: I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B > wrote: How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B > wrote: I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B > wrote: How to use E_DLG_STATE_CHANGED to identify the start of the call? On Wed, 20 Mar 2024 at 19:46, Ben Newlin > wrote: You can also use the REST client. And there are many other ways, as well. There is no single correct answer to the vague question of connecting to any generic “external application”. You must understand your systems and decide the best approach depending on the needs and capabilities of both the external application and OpenSIPS. Ben Newlin From: Users > on behalf of Bogdan-Andrei Iancu > Date: Wednesday, March 20, 2024 at 10:06 AM To: OpenSIPS users mailling list >, Prathibha B > Subject: Re: [OpenSIPS-Users] external applications EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Use the dialog events: https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED And you subscribe from outside OpenSIPS for such events: https://www.opensips.org/Documentation/Interface-Events-3-4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.03.2024 12:16, Prathibha B wrote: No. I want to pass START, CONNECT, END messages from OpenSIPS to external application. On Wed, 20 Mar 2024 at 15:42, Marcin Groszek > wrote: Well, to execute external command from opensips you may want to use EXEC module. this is a manual for v3.2: https://opensips.org/html/docs/modules/3.2.x/exec.html On 3/20/2024 5:00 AM, Prathibha B wrote: How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From sterlin at e2infosystems.com Thu Apr 11 10:58:22 2024 From: sterlin at e2infosystems.com (Sterlin Devanish) Date: Thu, 11 Apr 2024 16:28:22 +0530 Subject: [OpenSIPS-Users] Opensips as proxy for Asterisk Message-ID: Hi friends, I am new to opensips. I am working on handling Background calls for Flutter WebRTC clients using Asterisk. Since Asterisk doesn't support RFC8599, I am trying to configure opensips as a proxy server for Asterisk. I am using mid_registrar to forward the registration request from opensips to asterisk. It is perfectly working for SIP signaling, whereas for WebSockets the request is not reaching the asterisk from opensips. Kindly help me where I am going wrong, or help me handle this scenario. *Thanks,* *Sterlin Devanish D* -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 18 14:19:38 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 18 Apr 2024 16:19:38 +0200 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: No I don't. what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default value of net.core.rmem_max and net.core.rmem_default. Op do 18 apr 2024 om 16:02 schreef Ben Newlin : > Are you calling drop() anywhere in your script? > > > > https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 > > > > Ben Newlin > > > > *From: *Users on behalf of Johan De > Clercq > *Date: *Thursday, April 18, 2024 at 5:27 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] question on core statistics. > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > would it make sense to recompile with other flags ? And how do I set them > (I don't find these of menuconfig's compile options)? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > > > Can somebody explain also what both flags mean. > > > > Op do 18 apr 2024 om 11:07 schreef Johan De Clercq : > > would it make sense to recompile with other flags ? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > > > Can somebody explain also what both flags mean. > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > > > Op do 18 apr 2024 om 10:32 schreef Johan De Clercq : > > > > Guys, > > > > I have an opensips instance running with 24 worker children. > > The worker load is very low. > > UDP queues are on 50 megs. > > > > when i query via the OS > > cat /proc/net/udp > > sl local_address rem_address st tx_queue rx_queue tr tm->when > retrnsmt uid timeout inode ref pointer drops > > 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413684019 2 ffff880074820bc0 0 > > 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413766438 2 ffff880465e4a440 0 > > 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 412035865 2 ffff8803e5a56b80 0 > > 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26790 2 ffff88046c054840 0 > > 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26787 2 ffff88046c054bc0 0 > > 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26791 2 ffff88046c0544c0 0 > > 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15506 2 ffff88046dce5040 0 > > 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 22811 2 ffff880465e4ab40 0 > > 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20572 2 ffff88086d020800 0 > > 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20571 2 ffff88086d020b80 0 > > 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 100 0 17515 2 ffff8800368ac780 0 > > 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15505 2 ffff8800368acb00 0 > > > > => no drops > > > > what worries me is that there are drop requests and they go up when I > query via the mi interface > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 198107 > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 199157 > > opensipsctl_reg fifo get_statistics drop_requests > > core:drop_requests:: 204116 > > > > I don't see any memory issue, also the processload is low. > > > > > > so 3 questions: > > - what exactly is drop_request. > > - do I need to worry about this > > - how can I make them go lower. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 18 14:23:02 2024 From: Johan at democon.be (Johan De Clercq) Date: Thu, 18 Apr 2024 16:23:02 +0200 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: Wadii, this is the beginning of route[0] route { if (!mf_process_maxfwd_header("70") && $retcode==-1) { sl_send_reply("483","Too Many Hops"); xlog("callid [$ci] from [$fU] to [$tU] loop detected"); exit; }; force_rport(); t_on_failure("Trunk_On_Error"); if (has_totag()) { loose_route(); t_relay(); exit; } record_route(); create_dialog("B"); so I don't think that this is related ? Op do 18 apr 2024 om 16:12 schreef Wadii ELMAJDI | Evenmedia < wadii at evenmedia.fr>: > Calling exit() during the initial request and right before creating the > dialog also increments the drop_requests statistic > > > > *De :* Users *De la part de* Ben Newlin > *Envoyé :* jeudi 18 avril 2024 15:59 > *À :* OpenSIPS users mailling list > *Objet :* Re: [OpenSIPS-Users] question on core statistics. > > > > Are you calling drop() anywhere in your script? > > > > https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 > > > > Ben Newlin > > > > *From: *Users on behalf of Johan De > Clercq > *Date: *Thursday, April 18, 2024 at 5:27 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] question on core statistics. > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > would it make sense to recompile with other flags ? And how do I set them > (I don't find these of menuconfig's compile options)? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > > > Can somebody explain also what both flags mean. > > > > Op do 18 apr 2024 om 11:07 schreef Johan De Clercq : > > would it make sense to recompile with other flags ? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > > > Can somebody explain also what both flags mean. > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > > > Op do 18 apr 2024 om 10:32 schreef Johan De Clercq : > > > > Guys, > > > > I have an opensips instance running with 24 worker children. > > The worker load is very low. > > UDP queues are on 50 megs. > > > > when i query via the OS > > cat /proc/net/udp > > sl local_address rem_address st tx_queue rx_queue tr tm->when > retrnsmt uid timeout inode ref pointer drops > > 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413684019 2 ffff880074820bc0 0 > > 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 413766438 2 ffff880465e4a440 0 > > 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 412035865 2 ffff8803e5a56b80 0 > > 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26790 2 ffff88046c054840 0 > > 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26787 2 ffff88046c054bc0 0 > > 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 26791 2 ffff88046c0544c0 0 > > 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15506 2 ffff88046dce5040 0 > > 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 22811 2 ffff880465e4ab40 0 > > 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20572 2 ffff88086d020800 0 > > 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 20571 2 ffff88086d020b80 0 > > 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 100 0 17515 2 ffff8800368ac780 0 > > 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 > 00000000 0 0 15505 2 ffff8800368acb00 0 > > > > => no drops > > > > what worries me is that there are drop requests and they go up when I > query via the mi interface > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 198107 > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 199157 > > opensipsctl_reg fifo get_statistics drop_requests > > core:drop_requests:: 204116 > > > > I don't see any memory issue, also the processload is low. > > > > > > so 3 questions: > > - what exactly is drop_request. > > - do I need to worry about this > > - how can I make them go lower. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 18 14:29:41 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 18 Apr 2024 17:29:41 +0300 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: The `drop_requests` statistic is incremented when: * the request is dropped by a pre-script callback (like B2B when there is no script execution for certain messages) * the stateless `forward()` core function failed to send out something. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 18.04.2024 17:19, Johan De Clercq wrote: > No I don't. > what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default > value of net.core.rmem_max and net.core.rmem_default. > > Op do 18 apr 2024 om 16:02 schreef Ben Newlin : > > Are you calling drop() anywhere in your script? > > https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 > > Ben Newlin > > *From: *Users on behalf of > Johan De Clercq > *Date: *Thursday, April 18, 2024 at 5:27 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] question on core statistics. > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > ------------------------------------------------------------------------ > > would it make sense to recompile with other flags ? And how do I > set them  (I don't find these of menuconfig's compile options)? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > Can somebody explain also what both flags mean. > > Op do 18 apr 2024 om 11:07 schreef Johan De Clercq : > > would it make sense to recompile with other flags ? > > Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. > > Can somebody explain also what both flags mean. > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, > PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, > MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > Op do 18 apr 2024 om 10:32 schreef Johan De Clercq > : > > Guys, > > I have an opensips instance running with 24 worker children. > > The worker load is very low. > > UDP queues are on 50 megs. > > when i query via the OS > > cat /proc/net/udp > > sl  local_address rem_address   st tx_queue rx_queue tr > tm->when retrnsmt  uid  timeout inode ref pointer drops > > 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 413684019 2 > ffff880074820bc0 0 > > 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 413766438 2 > ffff880465e4a440 0 > > 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 412035865 2 > ffff8803e5a56b80 0 > > 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 26790 2 ffff88046c054840 0 > > 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 26787 2 ffff88046c054bc0 0 > > 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 26791 2 ffff88046c0544c0 0 > >  1972: 00000000:D92A 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 15506 2 ffff88046dce5040 0 > >  5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 22811 2 ffff880465e4ab40 0 > > 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 20572 2 ffff88086d020800 0 > > 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 20571 2 ffff88086d020b80 0 > > 13320: 00000000:857E 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  100        0 17515 2 ffff8800368ac780 0 > > 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 > 00:00000000 00000000  0        0 15505 2 ffff8800368acb00 0 > > => no drops > > what worries me is that there are drop requests and they  > go up when I query via the mi interface > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 198107 > > opensipsctl fifo get_statistics drop_requests > > core:drop_requests:: 199157 > > opensipsctl_reg fifo get_statistics drop_requests > > core:drop_requests:: 204116 > > I don't see any memory issue, also the processload is low. > > so 3 questions: > > - what exactly is drop_request. > > - do I need to worry about this > > - how can I make them go lower. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From brett at nemeroff.com Thu Apr 18 16:50:36 2024 From: brett at nemeroff.com (Brett Nemeroff) Date: Thu, 18 Apr 2024 11:50:36 -0500 Subject: [OpenSIPS-Users] Launch and async In-Reply-To: References: Message-ID: Can you do those operations outside of opensips? For example, can you receive that 200 and push an event to RabbitMQ and have an outboard worker do the delay and action? -Brett On Tue, Apr 16, 2024 at 9:27 AM Parthesh Bhavsar via Users < users at lists.opensips.org> wrote: > I'm facing a challenge in my OpenSIPS implementation where I need to > perform certain operations after receiving a 200 OK response, but only > after waiting for 10 seconds. Currently, I'm considering using the sleep() > function, but I'm concerned about its impact on performance. > > Is there a recommended approach to achieve this functionality without > compromising OpenSIPS performance? I'd appreciate any advice or alternative > solutions you can offer. > > > Regards, > > *Parthesh Bhavsar | Software Engineer | VOIP* > > > > On Tue, Apr 16, 2024 at 7:39 PM Parthesh Bhavsar < > parthesh.bhavsar at ecosmob.com> wrote: > >> Hello, >> I want to use Launch and async function for function sleep() as I need to >> wait for some time to do some operations but from the route which I use in >> above function I am not able to use any variable for my operation and from >> documentation it seems only able to use those variable which sleep() >> function use so is there any alternative solution to get variable??? or any >> other function on which I can wait for a specific time without blocking >> opensips?? >> >> >> Regards, >> >> *Parthesh Bhavsar | Software Engineer | VOIP* >> >> > > *[image: https://www.ecosmob.com/opensips-summit/] > * > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 19 04:34:14 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 19 Apr 2024 10:04:14 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: I am getting the error "Cannot handle call" On Thu, 11 Apr 2024 at 16:16, Prathibha B wrote: > I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. > Now getting the following errors in syslog: > > ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to > change URI Contact > ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change > URI Contact > ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was > sent out > ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm > ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, > [B2B.394.162.1712831164.913185781] > > On Thu, 11 Apr 2024 at 15:33, Prathibha B > wrote: > >> when I run opensips-cli -x mi cc_list_agents >> { >> "Agents": [ >> { >> "id": "101002", >> "Ref": 0, >> "Loged in": "NO" >> }, >> { >> "id": "101001", >> "Ref": 0, >> "Loged in": "NO" >> } >> ] >> } >> >> I've logged in 101001 and 101002 in the browser. But the Loged in status >> is No for both users. >> >> On Thu, 11 Apr 2024 at 15:10, Prathibha B >> wrote: >> >>> Cahnged it to $tU, Still getting error. >>> >>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>> wrote: >>> >>>> Hi, >>>> >>>> >>>> >>>> Something wrong about that part for me : ($(tU) == "112") >>>> >>>> Should be : ($tU == "112") >>>> >>>> >>>> >>>> Regards >>>> >>>> >>>> >>>> *De : *Users au nom de Prathibha B < >>>> prathibhab.tvm at gmail.com> >>>> *Répondre à : *OpenSIPS users mailling list >>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>> *À : *OpenSIPS users mailling list >>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>> >>>> >>>> >>>> bp.erss.in - asterisk >>>> >>>> bp.erss.in:1443 - opensips >>>> >>>> >>>> >>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>> wrote: >>>> >>>> *I've created the entries in cc_agents and cc_flows table.* >>>> >>>> >>>> >>>> *cc_agents table* >>>> >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> | id | agentid | location | logstate | msrp_location >>>> | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>> | 4 | support | 0 | 0 | >>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>> | 4 | support | 0 | 0 | >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> >>>> >>>> >>>> *cc_flows table* >>>> >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>> | message_flow_id | >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> | 1 | support | 256 | support | NULL | 0 | >>>> 0 | 0 | 0 | >>>> 0 | | 112 at bp.erss.in | NULL | NULL >>>> | >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> >>>> >>>> >>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>> wrote: >>>> >>>> Getting the following error in call center module >>>> >>>> >>>> >>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >>>> received) >>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>> >>>> >>>> >>>> Call center code in opensips.cfg: >>>> >>>> >>>> >>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>> if (!cc_handle_call("support")) { >>>> send_reply(403,"Cannot handle call"); >>>> exit; >>>> } >>>> >>>> } >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> _______________________________________________ Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From parthesh.bhavsar at ecosmob.com Fri Apr 19 04:52:57 2024 From: parthesh.bhavsar at ecosmob.com (Parthesh Bhavsar) Date: Fri, 19 Apr 2024 10:22:57 +0530 Subject: [OpenSIPS-Users] Launch and async In-Reply-To: References: Message-ID: Okay, I'll use some other way to do this wait and action requirement as sleep() are decreasing performance for longer periods of time. On Thu, 18 Apr, 2024, 22:20 Brett Nemeroff, wrote: > Can you do those operations outside of opensips? For example, can you > receive that 200 and push an event to RabbitMQ and have an outboard worker > do the delay and action? > -Brett > > > On Tue, Apr 16, 2024 at 9:27 AM Parthesh Bhavsar via Users < > users at lists.opensips.org> wrote: > >> I'm facing a challenge in my OpenSIPS implementation where I need to >> perform certain operations after receiving a 200 OK response, but only >> after waiting for 10 seconds. Currently, I'm considering using the sleep() >> function, but I'm concerned about its impact on performance. >> >> Is there a recommended approach to achieve this functionality without >> compromising OpenSIPS performance? I'd appreciate any advice or alternative >> solutions you can offer. >> >> >> Regards, >> >> *Parthesh Bhavsar | Software Engineer | VOIP* >> >> >> >> On Tue, Apr 16, 2024 at 7:39 PM Parthesh Bhavsar < >> parthesh.bhavsar at ecosmob.com> wrote: >> >>> Hello, >>> I want to use Launch and async function for function sleep() as I need >>> to wait for some time to do some operations but from the route which I use >>> in above function I am not able to use any variable for my operation and >>> from documentation it seems only able to use those variable which sleep() >>> function use so is there any alternative solution to get variable??? or any >>> other function on which I can wait for a specific time without blocking >>> opensips?? >>> >>> >>> Regards, >>> >>> *Parthesh Bhavsar | Software Engineer | VOIP* >>> >>> >> >> *[image: https://www.ecosmob.com/opensips-summit/] >> * >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 19 09:15:25 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 19 Apr 2024 14:45:25 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Can someone in this group help me with setting up a call center using opensips as the proxy server and asterisk as media server? On Fri, 19 Apr 2024 at 10:04, Prathibha B wrote: > I am getting the error "Cannot handle call" > > On Thu, 11 Apr 2024 at 16:16, Prathibha B > wrote: > >> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >> Now getting the following errors in syslog: >> >> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >> change URI Contact >> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >> change URI Contact >> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was >> sent out >> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >> [B2B.394.162.1712831164.913185781] >> >> On Thu, 11 Apr 2024 at 15:33, Prathibha B >> wrote: >> >>> when I run opensips-cli -x mi cc_list_agents >>> { >>> "Agents": [ >>> { >>> "id": "101002", >>> "Ref": 0, >>> "Loged in": "NO" >>> }, >>> { >>> "id": "101001", >>> "Ref": 0, >>> "Loged in": "NO" >>> } >>> ] >>> } >>> >>> I've logged in 101001 and 101002 in the browser. But the Loged in status >>> is No for both users. >>> >>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>> wrote: >>> >>>> Cahnged it to $tU, Still getting error. >>>> >>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>> wrote: >>>> >>>>> Hi, >>>>> >>>>> >>>>> >>>>> Something wrong about that part for me : ($(tU) == "112") >>>>> >>>>> Should be : ($tU == "112") >>>>> >>>>> >>>>> >>>>> Regards >>>>> >>>>> >>>>> >>>>> *De : *Users au nom de Prathibha B >>>>> >>>>> *Répondre à : *OpenSIPS users mailling list >>>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>>> *À : *OpenSIPS users mailling list >>>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>>> >>>>> >>>>> >>>>> bp.erss.in - asterisk >>>>> >>>>> bp.erss.in:1443 - opensips >>>>> >>>>> >>>>> >>>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>>> wrote: >>>>> >>>>> *I've created the entries in cc_agents and cc_flows table.* >>>>> >>>>> >>>>> >>>>> *cc_agents table* >>>>> >>>>> >>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>> | id | agentid | location | logstate | msrp_location >>>>> | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >>>>> >>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>>> | 4 | support | 0 | 0 | >>>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>>> | 4 | support | 0 | 0 | >>>>> >>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>> >>>>> >>>>> >>>>> *cc_flows table* >>>>> >>>>> >>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>>> | message_flow_id | >>>>> >>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>> | 1 | support | 256 | support | NULL | 0 | >>>>> 0 | 0 | 0 | >>>>> 0 | | 112 at bp.erss.in | NULL | NULL >>>>> | >>>>> >>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>> >>>>> >>>>> >>>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>>> wrote: >>>>> >>>>> Getting the following error in call center module >>>>> >>>>> >>>>> >>>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty >>>>> ID received) >>>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>>> >>>>> >>>>> >>>>> Call center code in opensips.cfg: >>>>> >>>>> >>>>> >>>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>>> if (!cc_handle_call("support")) { >>>>> send_reply(403,"Cannot handle call"); >>>>> exit; >>>>> } >>>>> >>>>> } >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Regards, >>>>> >>>>> B.Prathibha >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Regards, >>>>> >>>>> B.Prathibha >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Regards, >>>>> >>>>> B.Prathibha >>>>> >>>>> _______________________________________________ Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 19 11:07:02 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 19 Apr 2024 16:37:02 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: or freeswitch as the media server. On Fri, 19 Apr 2024 at 14:45, Prathibha B wrote: > Can someone in this group help me with setting up a call center using > opensips as the proxy server and asterisk as media server? > > On Fri, 19 Apr 2024 at 10:04, Prathibha B > wrote: > >> I am getting the error "Cannot handle call" >> >> On Thu, 11 Apr 2024 at 16:16, Prathibha B >> wrote: >> >>> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >>> Now getting the following errors in syslog: >>> >>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>> change URI Contact >>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>> change URI Contact >>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was >>> sent out >>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>> [B2B.394.162.1712831164.913185781] >>> >>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>> wrote: >>> >>>> when I run opensips-cli -x mi cc_list_agents >>>> { >>>> "Agents": [ >>>> { >>>> "id": "101002", >>>> "Ref": 0, >>>> "Loged in": "NO" >>>> }, >>>> { >>>> "id": "101001", >>>> "Ref": 0, >>>> "Loged in": "NO" >>>> } >>>> ] >>>> } >>>> >>>> I've logged in 101001 and 101002 in the browser. But the Loged in >>>> status is No for both users. >>>> >>>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>>> wrote: >>>> >>>>> Cahnged it to $tU, Still getting error. >>>>> >>>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>>> wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> >>>>>> >>>>>> Something wrong about that part for me : ($(tU) == "112") >>>>>> >>>>>> Should be : ($tU == "112") >>>>>> >>>>>> >>>>>> >>>>>> Regards >>>>>> >>>>>> >>>>>> >>>>>> *De : *Users au nom de Prathibha >>>>>> B >>>>>> *Répondre à : *OpenSIPS users mailling list >>>>> > >>>>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>>>> *À : *OpenSIPS users mailling list >>>>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>>>> >>>>>> >>>>>> >>>>>> bp.erss.in - asterisk >>>>>> >>>>>> bp.erss.in:1443 - opensips >>>>>> >>>>>> >>>>>> >>>>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>>>> wrote: >>>>>> >>>>>> *I've created the entries in cc_agents and cc_flows table.* >>>>>> >>>>>> >>>>>> >>>>>> *cc_agents table* >>>>>> >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> | id | agentid | location | logstate | >>>>>> msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time >>>>>> | >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>>>> | 4 | support | 0 | 0 | >>>>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>>>> | 4 | support | 0 | 0 | >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> >>>>>> >>>>>> >>>>>> *cc_flows table* >>>>>> >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>>>> | message_flow_id | >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> | 1 | support | 256 | support | NULL | 0 | >>>>>> 0 | 0 | 0 | >>>>>> 0 | | 112 at bp.erss.in | NULL | >>>>>> NULL | >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> >>>>>> >>>>>> >>>>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>>>> wrote: >>>>>> >>>>>> Getting the following error in call center module >>>>>> >>>>>> >>>>>> >>>>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty >>>>>> ID received) >>>>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>>>> >>>>>> >>>>>> >>>>>> Call center code in opensips.cfg: >>>>>> >>>>>> >>>>>> >>>>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>>>> if (!cc_handle_call("support")) { >>>>>> send_reply(403,"Cannot handle call"); >>>>>> exit; >>>>>> } >>>>>> >>>>>> } >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> _______________________________________________ Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> B.Prathibha >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From brett at nemeroff.com Fri Apr 19 13:38:51 2024 From: brett at nemeroff.com (Brett Nemeroff) Date: Fri, 19 Apr 2024 08:38:51 -0500 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: I don’t know the details, but I’d guess that because you have no agents logged in your handle call isn’t properly rewriting the URI. It may be useful to check the return status and log it from cc_handle_call. That may give you some insight on at least this stage of the call processing. -Brett On Fri, Apr 19, 2024 at 6:09 AM Prathibha B wrote: > or freeswitch as the media server. > > On Fri, 19 Apr 2024 at 14:45, Prathibha B > wrote: > >> Can someone in this group help me with setting up a call center using >> opensips as the proxy server and asterisk as media server? >> >> On Fri, 19 Apr 2024 at 10:04, Prathibha B >> wrote: >> >>> I am getting the error "Cannot handle call" >>> >>> On Thu, 11 Apr 2024 at 16:16, Prathibha B >>> wrote: >>> >>>> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >>>> Now getting the following errors in syslog: >>>> >>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>> change URI Contact >>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>> change URI Contact >>>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 >>>> was sent out >>>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>>> [B2B.394.162.1712831164.913185781] >>>> >>>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>>> wrote: >>>> >>>>> when I run opensips-cli -x mi cc_list_agents >>>>> { >>>>> "Agents": [ >>>>> { >>>>> "id": "101002", >>>>> "Ref": 0, >>>>> "Loged in": "NO" >>>>> }, >>>>> { >>>>> "id": "101001", >>>>> "Ref": 0, >>>>> "Loged in": "NO" >>>>> } >>>>> ] >>>>> } >>>>> >>>>> I've logged in 101001 and 101002 in the browser. But the Loged in >>>>> status is No for both users. >>>>> >>>>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>>>> wrote: >>>>> >>>>>> Cahnged it to $tU, Still getting error. >>>>>> >>>>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>>>> wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> >>>>>>> >>>>>>> Something wrong about that part for me : ($(tU) == "112") >>>>>>> >>>>>>> Should be : ($tU == "112") >>>>>>> >>>>>>> >>>>>>> >>>>>>> Regards >>>>>>> >>>>>>> >>>>>>> >>>>>>> *De : *Users au nom de Prathibha >>>>>>> B >>>>>>> *Répondre à : *OpenSIPS users mailling list < >>>>>>> users at lists.opensips.org> >>>>>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>>>>> *À : *OpenSIPS users mailling list >>>>>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>>>>> >>>>>>> >>>>>>> >>>>>>> bp.erss.in - asterisk >>>>>>> >>>>>>> bp.erss.in:1443 - opensips >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> *I've created the entries in cc_agents and cc_flows table.* >>>>>>> >>>>>>> >>>>>>> >>>>>>> *cc_agents table* >>>>>>> >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> | id | agentid | location | logstate | >>>>>>> msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time >>>>>>> | >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>>>>> | 4 | support | 0 | 0 | >>>>>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>>>>> | 4 | support | 0 | 0 | >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> >>>>>>> >>>>>>> >>>>>>> *cc_flows table* >>>>>>> >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>>>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>>>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>>>>> | message_flow_id | >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> | 1 | support | 256 | support | NULL | 0 | >>>>>>> 0 | 0 | 0 | >>>>>>> 0 | | 112 at bp.erss.in | NULL | >>>>>>> NULL | >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> Getting the following error in call center module >>>>>>> >>>>>>> >>>>>>> >>>>>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty >>>>>>> ID received) >>>>>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>>>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>>>>> >>>>>>> >>>>>>> >>>>>>> Call center code in opensips.cfg: >>>>>>> >>>>>>> >>>>>>> >>>>>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>>>>> if (!cc_handle_call("support")) { >>>>>>> send_reply(403,"Cannot handle call"); >>>>>>> exit; >>>>>>> } >>>>>>> >>>>>>> } >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> _______________________________________________ Users mailing list >>>>>>> Users at lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users at lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> B.Prathibha >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 19 13:43:17 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 19 Apr 2024 13:43:17 +0000 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Ok. Sent from Outlook for Android ________________________________ From: Users on behalf of Brett Nemeroff Sent: Friday, April 19, 2024 7:08:51 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Call Center error I don’t know the details, but I’d guess that because you have no agents logged in your handle call isn’t properly rewriting the URI. It may be useful to check the return status and log it from cc_handle_call. That may give you some insight on at least this stage of the call processing. -Brett On Fri, Apr 19, 2024 at 6:09 AM Prathibha B > wrote: or freeswitch as the media server. On Fri, 19 Apr 2024 at 14:45, Prathibha B > wrote: Can someone in this group help me with setting up a call center using opensips as the proxy server and asterisk as media server? On Fri, 19 Apr 2024 at 10:04, Prathibha B > wrote: I am getting the error "Cannot handle call" On Thu, 11 Apr 2024 at 16:16, Prathibha B > wrote: I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. Now getting the following errors in syslog: ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change URI Contact ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change URI Contact ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was sent out ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, [B2B.394.162.1712831164.913185781] On Thu, 11 Apr 2024 at 15:33, Prathibha B > wrote: when I run opensips-cli -x mi cc_list_agents { "Agents": [ { "id": "101002", "Ref": 0, "Loged in": "NO" }, { "id": "101001", "Ref": 0, "Loged in": "NO" } ] } I've logged in 101001 and 101002 in the browser. But the Loged in status is No for both users. On Thu, 11 Apr 2024 at 15:10, Prathibha B > wrote: Cahnged it to $tU, Still getting error. On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent > wrote: Hi, Something wrong about that part for me : ($(tU) == "112") Should be : ($tU == "112") Regards De : Users > au nom de Prathibha B > Répondre à : OpenSIPS users mailling list > Date : jeudi 11 avril 2024 à 11:03 À : OpenSIPS users mailling list > Objet : Re: [OpenSIPS-Users] Call Center error bp.erss.in - asterisk bp.erss.in:1443 - opensips On Thu, 11 Apr 2024 at 14:29, Prathibha B > wrote: I've created the entries in cc_agents and cc_flows table. cc_agents table +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | id | agentid | location | logstate | msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | 4 | support | 0 | 0 | +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ cc_flows table +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | id | flowid | priority | skill | prependcid | max_wrapup_time | dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | dissuading_qsize_th | message_welcome | message_queue | message_dissuading | message_flow_id | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ | 1 | support | 256 | support | NULL | 0 | 0 | 0 | 0 | 0 | | 112 at bp.erss.in | NULL | NULL | +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ On Thu, 11 Apr 2024 at 14:27, Prathibha B > wrote: Getting the following error in call center module ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: ERROR:call_center:w_handle_call: failed to set new destination for call Call center code in opensips.cfg: if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { if (!cc_handle_call("support")) { send_reply(403,"Cannot handle call"); exit; } } -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha -- Regards, B.Prathibha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Apr 19 08:13:25 2024 From: johan at democon.be (johan) Date: Fri, 19 Apr 2024 10:13:25 +0200 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: <3aa2b48f-fa7d-4b66-ae4a-55fffc139409@democon.be> Bogdan, on the augmenting drop_requests, drop is not used in that script.  Neither is forward.  Everything is t_relay.  There is no b2b. The thing is that we observed drops on udp level. We followed the recommendations of  Ovidiu Sas's presentation of last year in Houston: - increase PKG mem - increase SHM mem - increase workers to 24 so that the queue empties faster. - we checked the udp queues on linux level and we saw drops there.     => we augmented them to 50 megs (sysctl -w net.core.rmem_max=52428800 and sysctl -w net.core.rmem_default=52428800) and the drops on OS level where gone. Also worker and memory load are max 30 %. Hence we thought that we were okay, but still drops on opensips level.  Net result was that this node in the system lost all connection with the destination of the loadbalancer although it received keep alive options responses from the loadbalancer destination on its NIC (we could see that in a continuously running tcpdump). => hence it seems that it is opensips's receive buffer that is too small (as I read  the description : "Returns the number of requests dropped even before entering the script routing logic.", I thought that this pointed to the receive buffer of opensips).  All of this is happening on a physical machine on which two other opensips instances are running also. Interestingly enough the problem is only observed in the instance that handles registrations and invites (1600 REG/ s and 300 INV /s). Therefore we dived a bit deeper and came on this MAX_RECV_BUFFER_SIZE 262144 (which is the default udp queue size setting on linux).  Could this be related somehow ? Secondly, what would the recommendation be for scaling a system like this ? On 18/04/2024 16:29, Bogdan-Andrei Iancu wrote: > The `drop_requests` statistic is incremented when: > * the request is dropped by a pre-script callback (like B2B when there > is no script execution for certain messages) > * the stateless `forward()` core function failed to send out something. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 18.04.2024 17:19, Johan De Clercq wrote: >> No I don't. >> what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the >> default value of net.core.rmem_max and net.core.rmem_default. >> >> Op do 18 apr 2024 om 16:02 schreef Ben Newlin : >> >> Are you calling drop() anywhere in your script? >> >> https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 >> >> Ben Newlin >> >> *From: *Users on behalf of >> Johan De Clercq >> *Date: *Thursday, April 18, 2024 at 5:27 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] question on core statistics. >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> ------------------------------------------------------------------------ >> >> would it make sense to recompile with other flags ? And how do I >> set them  (I don't find these of menuconfig's compile options)? >> >> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. >> >> Can somebody explain also what both flags mean. >> >> Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >> : >> >> would it make sense to recompile with other flags ? >> >> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. >> >> Can somebody explain also what both flags mean. >> >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, >> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >> >> poll method support: poll, epoll, sigio_rt, select. >> >> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >> : >> >> Guys, >> >> I have an opensips instance running with 24 worker children. >> >> The worker load is very low. >> >> UDP queues are on 50 megs. >> >> when i query via the OS >> >> cat /proc/net/udp >> >> sl  local_address rem_address   st tx_queue rx_queue tr >> tm->when retrnsmt  uid  timeout inode ref pointer drops >> >> 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 413684019 2 >> ffff880074820bc0 0 >> >> 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 413766438 2 >> ffff880465e4a440 0 >> >> 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 412035865 2 >> ffff8803e5a56b80 0 >> >> 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 26790 2 >> ffff88046c054840 0 >> >> 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 26787 2 >> ffff88046c054bc0 0 >> >> 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 26791 2 >> ffff88046c0544c0 0 >> >>  1972: 00000000:D92A 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 15506 2 >> ffff88046dce5040 0 >> >>  5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 22811 2 >> ffff880465e4ab40 0 >> >> 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 20572 2 >> ffff88086d020800 0 >> >> 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 20571 2 >> ffff88086d020b80 0 >> >> 13320: 00000000:857E 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000  100        0 17515 2 >> ffff8800368ac780 0 >> >> 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000    0        0 15505 2 >> ffff8800368acb00 0 >> >> => no drops >> >> what worries me is that there are drop requests and they  >> go up when I query via the mi interface >> >> opensipsctl fifo get_statistics drop_requests >> >> core:drop_requests:: 198107 >> >> opensipsctl fifo get_statistics drop_requests >> >> core:drop_requests:: 199157 >> >> opensipsctl_reg fifo get_statistics drop_requests >> >> core:drop_requests:: 204116 >> >> I don't see any memory issue, also the processload is low. >> >> so 3 questions: >> >> - what exactly is drop_request. >> >> - do I need to worry about this >> >> - how can I make them go lower. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 19 15:14:34 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 19 Apr 2024 18:14:34 +0300 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: <3aa2b48f-fa7d-4b66-ae4a-55fffc139409@democon.be> References: <3aa2b48f-fa7d-4b66-ae4a-55fffc139409@democon.be> Message-ID: <84023e1a-e6c1-4ef7-b348-b3adc0fad9c5@opensips.org> Somehow I think there is a confusion - the drop_requests stat has nothing to do with the dropping on the socket buffers (net level). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 19.04.2024 11:13, johan wrote: > > Bogdan, > > on the augmenting drop_requests, > > drop is not used in that script.  Neither is forward. Everything is > t_relay.  There is no b2b. > > The thing is that we observed drops on udp level. > > We followed the recommendations of  Ovidiu Sas's presentation of last > year in Houston: > > - increase PKG mem > > - increase SHM mem > > - increase workers to 24 so that the queue empties faster. > > - we checked the udp queues on linux level and we saw drops there. > >     => we augmented them to 50 megs (sysctl -w > net.core.rmem_max=52428800 and sysctl -w > net.core.rmem_default=52428800) and the drops on OS level where gone. > > Also worker and memory load are max 30 %. > > > Hence we thought that we were okay, but still drops on opensips > level.  Net result was that this node in the system lost all > connection with the destination of the loadbalancer although it > received keep alive options responses from the loadbalancer > destination on its NIC (we could see that in a continuously running > tcpdump). > > => hence it seems that it is opensips's receive buffer that is too > small (as I read  the description : "Returns the number of requests > dropped even before entering the script routing logic.", I thought > that this pointed to the receive buffer of opensips).  All of this is > happening on a physical machine on which two other opensips instances > are running also. Interestingly enough the problem is only observed in > the instance that handles registrations and invites (1600 REG/ s and > 300 INV /s). > > > Therefore we dived a bit deeper and came on this MAX_RECV_BUFFER_SIZE > 262144 (which is the default udp queue size setting on linux).  Could > this be related somehow ? > > > Secondly, what would the recommendation be for scaling a system like > this ? > > > > > > > > > On 18/04/2024 16:29, Bogdan-Andrei Iancu wrote: >> The `drop_requests` statistic is incremented when: >> * the request is dropped by a pre-script callback (like B2B when >> there is no script execution for certain messages) >> * the stateless `forward()` core function failed to send out something. >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> On 18.04.2024 17:19, Johan De Clercq wrote: >>> No I don't. >>> what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the >>> default value of net.core.rmem_max and net.core.rmem_default. >>> >>> Op do 18 apr 2024 om 16:02 schreef Ben Newlin : >>> >>> Are you calling drop() anywhere in your script? >>> >>> https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 >>> >>> Ben Newlin >>> >>> *From: *Users on behalf of >>> Johan De Clercq >>> *Date: *Thursday, April 18, 2024 at 5:27 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] question on core statistics. >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> ------------------------------------------------------------------------ >>> >>> would it make sense to recompile with other flags ? And how do I >>> set them  (I don't find these of menuconfig's compile options)? >>> >>> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. >>> >>> Can somebody explain also what both flags mean. >>> >>> Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >>> : >>> >>> would it make sense to recompile with other flags ? >>> >>> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE >>> 65535. >>> >>> Can somebody explain also what both flags mean. >>> >>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, >>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>> >>> poll method support: poll, epoll, sigio_rt, select. >>> >>> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >>> : >>> >>> Guys, >>> >>> I have an opensips instance running with 24 worker >>> children. >>> >>> The worker load is very low. >>> >>> UDP queues are on 50 megs. >>> >>> when i query via the OS >>> >>> cat /proc/net/udp >>> >>>   sl  local_address rem_address   st tx_queue rx_queue >>> tr tm->when retrnsmt   uid timeout inode ref pointer drops >>> >>>   590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 413684019 2 >>> ffff880074820bc0 0 >>> >>>   591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 413766438 2 >>> ffff880465e4a440 0 >>> >>>   592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 412035865 2 >>> ffff8803e5a56b80 0 >>> >>>   934: 01231D0A:151C 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 26790 2 ffff88046c054840 0 >>> >>>   935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 26787 2 ffff88046c054bc0 0 >>> >>>   935: 01231D0A:151D 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 26791 2 ffff88046c0544c0 0 >>> >>>  1972: 00000000:D92A 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 15506 2 ffff88046dce5040 0 >>> >>>  5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 22811 2 ffff880465e4ab40 0 >>> >>> 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 20572 2 ffff88086d020800 0 >>> >>> 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 20571 2 ffff88086d020b80 0 >>> >>> 13320: 00000000:857E 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000   100   0 17515 2 ffff8800368ac780 0 >>> >>> 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 >>> 00:00000000 00000000     0   0 15505 2 ffff8800368acb00 0 >>> >>> => no drops >>> >>> what worries me is that there are drop requests and >>> they  go up when I query via the mi interface >>> >>> opensipsctl fifo get_statistics drop_requests >>> >>> core:drop_requests:: 198107 >>> >>> opensipsctl fifo get_statistics drop_requests >>> >>> core:drop_requests:: 199157 >>> >>> opensipsctl_reg fifo get_statistics drop_requests >>> >>> core:drop_requests:: 204116 >>> >>> I don't see any memory issue, also the processload is low. >>> >>> so 3 questions: >>> >>> - what exactly is drop_request. >>> >>> - do I need to worry about this >>> >>> - how can I make them go lower. >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 20 08:47:13 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 20 Apr 2024 14:17:13 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: I am getting 401 Unauthorized error. How to resolve this issue? On Fri, 19 Apr 2024 at 19:13, Prathibha B wrote: > Ok. > > Sent from Outlook for Android > ------------------------------ > *From:* Users on behalf of Brett > Nemeroff > *Sent:* Friday, April 19, 2024 7:08:51 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] Call Center error > > I don’t know the details, but I’d guess that because you have no agents > logged in your handle call isn’t properly rewriting the URI. It may be > useful to check the return status and log it from cc_handle_call. That may > give you some insight on at least this stage of the call processing. > > -Brett > > > On Fri, Apr 19, 2024 at 6:09 AM Prathibha B > wrote: > > or freeswitch as the media server. > > On Fri, 19 Apr 2024 at 14:45, Prathibha B > wrote: > > Can someone in this group help me with setting up a call center using > opensips as the proxy server and asterisk as media server? > > On Fri, 19 Apr 2024 at 10:04, Prathibha B > wrote: > > I am getting the error "Cannot handle call" > > On Thu, 11 Apr 2024 at 16:16, Prathibha B > wrote: > > I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. > Now getting the following errors in syslog: > > ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to > change URI Contact > ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change > URI Contact > ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was > sent out > ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm > ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, > [B2B.394.162.1712831164.913185781] > > On Thu, 11 Apr 2024 at 15:33, Prathibha B > wrote: > > when I run opensips-cli -x mi cc_list_agents > { > "Agents": [ > { > "id": "101002", > "Ref": 0, > "Loged in": "NO" > }, > { > "id": "101001", > "Ref": 0, > "Loged in": "NO" > } > ] > } > > I've logged in 101001 and 101002 in the browser. But the Loged in status > is No for both users. > > On Thu, 11 Apr 2024 at 15:10, Prathibha B > wrote: > > Cahnged it to $tU, Still getting error. > > On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent > wrote: > > Hi, > > > > Something wrong about that part for me : ($(tU) == "112") > > Should be : ($tU == "112") > > > > Regards > > > > *De : *Users au nom de Prathibha B < > prathibhab.tvm at gmail.com> > *Répondre à : *OpenSIPS users mailling list > *Date : *jeudi 11 avril 2024 à 11:03 > *À : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] Call Center error > > > > bp.erss.in - asterisk > > bp.erss.in:1443 - opensips > > > > On Thu, 11 Apr 2024 at 14:29, Prathibha B > wrote: > > *I've created the entries in cc_agents and cc_flows table.* > > > > *cc_agents table* > > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | id | agentid | location | logstate | msrp_location | > msrp_max_sessions | skills | wrapup_end_time | wrapup_time | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | > 4 | support | 0 | 0 | > > +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ > > > > *cc_flows table* > > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | id | flowid | priority | skill | prependcid | max_wrapup_time | > dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | > dissuading_qsize_th | message_welcome | message_queue | message_dissuading > | message_flow_id | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > | 1 | support | 256 | support | NULL | 0 | > 0 | 0 | 0 | > 0 | | 112 at bp.erss.in | NULL | NULL > | > > +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ > > > > On Thu, 11 Apr 2024 at 14:27, Prathibha B > wrote: > > Getting the following error in call center module > > > > ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID > received) > Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: > ERROR:call_center:w_handle_call: failed to set new destination for call > > > > Call center code in opensips.cfg: > > > > if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { > if (!cc_handle_call("support")) { > send_reply(403,"Cannot handle call"); > exit; > } > > } > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > > > > -- > > Regards, > > B.Prathibha > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Regards, > B.Prathibha > > > > -- > Regards, > B.Prathibha > > > > -- > Regards, > B.Prathibha > > > > -- > Regards, > B.Prathibha > > > > -- > Regards, > B.Prathibha > > > > -- > Regards, > B.Prathibha > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.royannais at orange.com Sat Apr 20 08:50:35 2024 From: julien.royannais at orange.com (julien.royannais at orange.com) Date: Sat, 20 Apr 2024 08:50:35 +0000 Subject: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic Message-ID: Hello everyone, I'm reaching out to get your opinion on using openSIPS with a business Application Server (AS) that would handle the routing logic. At first glance, the SEAS module seems designed for this purpose, but it doesn't appear to be a good fit as it seems highly coupled with a Sip Servlet implementation using a specific protocol only supported for WeSIP. It might be better to use REST interfaces, a 302 redirect-based mechanism, or possibly another module. Thank you for your insights & advice! JR Orange Restricted Orange Restricted ____________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladpaiu at opensips.org Sat Apr 20 10:07:47 2024 From: vladpaiu at opensips.org (Vlad Paiu) Date: Sat, 20 Apr 2024 13:07:47 +0300 Subject: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic In-Reply-To: References: Message-ID: Hello, Since OpenSIPS is a SIP server, I would try to leverage it's capabilities and communicate over SIP with the AS - that means scripting your way through. On 20.04.2024 11:50, julien.royannais at orange.com wrote: > > Hello everyone, > > I'm reaching out to get your opinion on using openSIPS with a business > Application Server (AS) that would handle the routing logic. > > At first glance, the SEAS module seems designed for this purpose, but > it doesn't appear to be a good fit as it seems highly coupled with a > Sip Servlet implementation using a specific protocol only supported > for WeSIP. > > It might be better to use REST interfaces, a 302 redirect-based > mechanism, or possibly another module. > > Thank you for your insights & advice! > > JR > > Orange Restricted > > > Orange Restricted > > ____________________________________________________________________________________________________________ > Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler > a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, > Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. > > This message and its attachments may contain confidential or privileged information that may be protected by law; > they should not be distributed, used or copied without authorisation. > If you have received this email in error, please notify the sender and delete this message and its attachments. > As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. > Thank you. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 20 15:12:19 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 20 Apr 2024 20:42:19 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: A non stop music is getting played but the call is not getting redirected to the call center agent. On Sat, 20 Apr 2024 at 14:17, Prathibha B wrote: > I am getting 401 Unauthorized error. How to resolve this issue? > > On Fri, 19 Apr 2024 at 19:13, Prathibha B > wrote: > >> Ok. >> >> Sent from Outlook for Android >> ------------------------------ >> *From:* Users on behalf of Brett >> Nemeroff >> *Sent:* Friday, April 19, 2024 7:08:51 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] Call Center error >> >> I don’t know the details, but I’d guess that because you have no agents >> logged in your handle call isn’t properly rewriting the URI. It may be >> useful to check the return status and log it from cc_handle_call. That may >> give you some insight on at least this stage of the call processing. >> >> -Brett >> >> >> On Fri, Apr 19, 2024 at 6:09 AM Prathibha B >> wrote: >> >> or freeswitch as the media server. >> >> On Fri, 19 Apr 2024 at 14:45, Prathibha B >> wrote: >> >> Can someone in this group help me with setting up a call center using >> opensips as the proxy server and asterisk as media server? >> >> On Fri, 19 Apr 2024 at 10:04, Prathibha B >> wrote: >> >> I am getting the error "Cannot handle call" >> >> On Thu, 11 Apr 2024 at 16:16, Prathibha B >> wrote: >> >> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >> Now getting the following errors in syslog: >> >> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >> change URI Contact >> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >> change URI Contact >> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was >> sent out >> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >> [B2B.394.162.1712831164.913185781] >> >> On Thu, 11 Apr 2024 at 15:33, Prathibha B >> wrote: >> >> when I run opensips-cli -x mi cc_list_agents >> { >> "Agents": [ >> { >> "id": "101002", >> "Ref": 0, >> "Loged in": "NO" >> }, >> { >> "id": "101001", >> "Ref": 0, >> "Loged in": "NO" >> } >> ] >> } >> >> I've logged in 101001 and 101002 in the browser. But the Loged in status >> is No for both users. >> >> On Thu, 11 Apr 2024 at 15:10, Prathibha B >> wrote: >> >> Cahnged it to $tU, Still getting error. >> >> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >> wrote: >> >> Hi, >> >> >> >> Something wrong about that part for me : ($(tU) == "112") >> >> Should be : ($tU == "112") >> >> >> >> Regards >> >> >> >> *De : *Users au nom de Prathibha B < >> prathibhab.tvm at gmail.com> >> *Répondre à : *OpenSIPS users mailling list >> *Date : *jeudi 11 avril 2024 à 11:03 >> *À : *OpenSIPS users mailling list >> *Objet : *Re: [OpenSIPS-Users] Call Center error >> >> >> >> bp.erss.in - asterisk >> >> bp.erss.in:1443 - opensips >> >> >> >> On Thu, 11 Apr 2024 at 14:29, Prathibha B >> wrote: >> >> *I've created the entries in cc_agents and cc_flows table.* >> >> >> >> *cc_agents table* >> >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | id | agentid | location | logstate | msrp_location | >> msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL | >> 4 | support | 0 | 0 | >> >> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >> >> >> >> *cc_flows table* >> >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | id | flowid | priority | skill | prependcid | max_wrapup_time | >> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >> | message_flow_id | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> | 1 | support | 256 | support | NULL | 0 | >> 0 | 0 | 0 | >> 0 | | 112 at bp.erss.in | NULL | NULL >> | >> >> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >> >> >> >> On Thu, 11 Apr 2024 at 14:27, Prathibha B >> wrote: >> >> Getting the following error in call center module >> >> >> >> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >> received) >> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >> ERROR:call_center:w_handle_call: failed to set new destination for call >> >> >> >> Call center code in opensips.cfg: >> >> >> >> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >> if (!cc_handle_call("support")) { >> send_reply(403,"Cannot handle call"); >> exit; >> } >> >> } >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> >> >> >> -- >> >> Regards, >> >> B.Prathibha >> >> _______________________________________________ Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Regards, >> B.Prathibha >> >> >> >> -- >> Regards, >> B.Prathibha >> >> >> >> -- >> Regards, >> B.Prathibha >> >> >> >> -- >> Regards, >> B.Prathibha >> >> >> >> -- >> Regards, >> B.Prathibha >> >> >> >> -- >> Regards, >> B.Prathibha >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Sat Apr 20 15:21:45 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sat, 20 Apr 2024 20:51:45 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: when I run opensips-cli -x mi cc_list_calls { "Calls": [ { "id": "219.3", "Ref": 2, "State": "queued", "Call Time": 187, "Flow": "112" }, { "id": "219.2", "Ref": 2, "State": "queued", "Call Time": 187, "Flow": "112" }, { "id": "219.1", "Ref": 2, "State": "queued", "Call Time": 758, "Flow": "112" }, { "id": "219.0", "Ref": 2, "State": "queued", "Call Time": 758, "Flow": "112" } ] } opensips-cli -x mi cc_list_agents { "Agents": [ { "id": "101003", "Ref": 0, "Loged in": "YES", "State": "wrapup", "Wrapup-ends": 16 }, { "id": "101002", "Ref": 0, "Loged in": "YES", "State": "wrapup", "Wrapup-ends": 27 }, { "id": "101001", "Ref": 0, "Loged in": "YES", "State": "wrapup", "Wrapup-ends": 29 } ] } On Sat, 20 Apr 2024 at 20:42, Prathibha B wrote: > A non stop music is getting played but the call is not getting redirected > to the call center agent. > > On Sat, 20 Apr 2024 at 14:17, Prathibha B > wrote: > >> I am getting 401 Unauthorized error. How to resolve this issue? >> >> On Fri, 19 Apr 2024 at 19:13, Prathibha B >> wrote: >> >>> Ok. >>> >>> Sent from Outlook for Android >>> ------------------------------ >>> *From:* Users on behalf of Brett >>> Nemeroff >>> *Sent:* Friday, April 19, 2024 7:08:51 PM >>> *To:* OpenSIPS users mailling list >>> *Subject:* Re: [OpenSIPS-Users] Call Center error >>> >>> I don’t know the details, but I’d guess that because you have no agents >>> logged in your handle call isn’t properly rewriting the URI. It may be >>> useful to check the return status and log it from cc_handle_call. That may >>> give you some insight on at least this stage of the call processing. >>> >>> -Brett >>> >>> >>> On Fri, Apr 19, 2024 at 6:09 AM Prathibha B >>> wrote: >>> >>> or freeswitch as the media server. >>> >>> On Fri, 19 Apr 2024 at 14:45, Prathibha B >>> wrote: >>> >>> Can someone in this group help me with setting up a call center using >>> opensips as the proxy server and asterisk as media server? >>> >>> On Fri, 19 Apr 2024 at 10:04, Prathibha B >>> wrote: >>> >>> I am getting the error "Cannot handle call" >>> >>> On Thu, 11 Apr 2024 at 16:16, Prathibha B >>> wrote: >>> >>> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >>> Now getting the following errors in syslog: >>> >>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>> change URI Contact >>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>> change URI Contact >>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was >>> sent out >>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>> [B2B.394.162.1712831164.913185781] >>> >>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>> wrote: >>> >>> when I run opensips-cli -x mi cc_list_agents >>> { >>> "Agents": [ >>> { >>> "id": "101002", >>> "Ref": 0, >>> "Loged in": "NO" >>> }, >>> { >>> "id": "101001", >>> "Ref": 0, >>> "Loged in": "NO" >>> } >>> ] >>> } >>> >>> I've logged in 101001 and 101002 in the browser. But the Loged in status >>> is No for both users. >>> >>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>> wrote: >>> >>> Cahnged it to $tU, Still getting error. >>> >>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>> wrote: >>> >>> Hi, >>> >>> >>> >>> Something wrong about that part for me : ($(tU) == "112") >>> >>> Should be : ($tU == "112") >>> >>> >>> >>> Regards >>> >>> >>> >>> *De : *Users au nom de Prathibha B < >>> prathibhab.tvm at gmail.com> >>> *Répondre à : *OpenSIPS users mailling list >>> *Date : *jeudi 11 avril 2024 à 11:03 >>> *À : *OpenSIPS users mailling list >>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>> >>> >>> >>> bp.erss.in - asterisk >>> >>> bp.erss.in:1443 - opensips >>> >>> >>> >>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>> wrote: >>> >>> *I've created the entries in cc_agents and cc_flows table.* >>> >>> >>> >>> *cc_agents table* >>> >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> | id | agentid | location | logstate | msrp_location | >>> msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>> | 4 | support | 0 | 0 | >>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>> | 4 | support | 0 | 0 | >>> >>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>> >>> >>> >>> *cc_flows table* >>> >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>> | message_flow_id | >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> | 1 | support | 256 | support | NULL | 0 | >>> 0 | 0 | 0 | >>> 0 | | 112 at bp.erss.in | NULL | NULL >>> | >>> >>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>> >>> >>> >>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>> wrote: >>> >>> Getting the following error in call center module >>> >>> >>> >>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >>> received) >>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>> ERROR:call_center:w_handle_call: failed to set new destination for call >>> >>> >>> >>> Call center code in opensips.cfg: >>> >>> >>> >>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>> if (!cc_handle_call("support")) { >>> send_reply(403,"Cannot handle call"); >>> exit; >>> } >>> >>> } >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> >>> >>> >>> -- >>> >>> Regards, >>> >>> B.Prathibha >>> >>> _______________________________________________ Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Mon Apr 22 06:17:12 2024 From: johan at democon.be (Johan De Clercq) Date: Mon, 22 Apr 2024 06:17:12 +0000 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: Goed morning, How can I then increase opensips’s internal queue size? Best regards, Johan Verzonden vanuit Outlook voor iOS ________________________________ Van: Bogdan-Andrei Iancu Verzonden: Thursday, April 18, 2024 4:29:41 PM Aan: OpenSIPS users mailling list ; Johan De Clercq Onderwerp: Re: [OpenSIPS-Users] question on core statistics. The `drop_requests` statistic is incremented when: * the request is dropped by a pre-script callback (like B2B when there is no script execution for certain messages) * the stateless `forward()` core function failed to send out something. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 18.04.2024 17:19, Johan De Clercq wrote: No I don't. what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default value of net.core.rmem_max and net.core.rmem_default. Op do 18 apr 2024 om 16:02 schreef Ben Newlin >: Are you calling drop() anywhere in your script? https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 Ben Newlin From: Users > on behalf of Johan De Clercq > Date: Thursday, April 18, 2024 at 5:27 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] question on core statistics. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ would it make sense to recompile with other flags ? And how do I set them (I don't find these of menuconfig's compile options)? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >: would it make sense to recompile with other flags ? Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. Can somebody explain also what both flags mean. flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >: Guys, I have an opensips instance running with 24 worker children. The worker load is very low. UDP queues are on 50 megs. when i query via the OS cat /proc/net/udp sl local_address rem_address st tx_queue rx_queue tr tm->when retrnsmt uid timeout inode ref pointer drops 590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413684019 2 ffff880074820bc0 0 591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 413766438 2 ffff880465e4a440 0 592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 412035865 2 ffff8803e5a56b80 0 934: 01231D0A:151C 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26790 2 ffff88046c054840 0 935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26787 2 ffff88046c054bc0 0 935: 01231D0A:151D 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 26791 2 ffff88046c0544c0 0 1972: 00000000:D92A 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15506 2 ffff88046dce5040 0 5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 22811 2 ffff880465e4ab40 0 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20572 2 ffff88086d020800 0 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 20571 2 ffff88086d020b80 0 13320: 00000000:857E 00000000:0000 07 00000000:00000000 00:00000000 00000000 100 0 17515 2 ffff8800368ac780 0 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 00:00000000 00000000 0 0 15505 2 ffff8800368acb00 0 => no drops what worries me is that there are drop requests and they go up when I query via the mi interface opensipsctl fifo get_statistics drop_requests core:drop_requests:: 198107 opensipsctl fifo get_statistics drop_requests core:drop_requests:: 199157 opensipsctl_reg fifo get_statistics drop_requests core:drop_requests:: 204116 I don't see any memory issue, also the processload is low. so 3 questions: - what exactly is drop_request. - do I need to worry about this - how can I make them go lower. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From devang.dhandhalya at ecosmob.com Mon Apr 22 11:50:55 2024 From: devang.dhandhalya at ecosmob.com (Devang Dhandhalya) Date: Mon, 22 Apr 2024 17:20:55 +0530 Subject: [OpenSIPS-Users] Ms Teams to SBC call not working Message-ID: Hello All, I Configured Direct routing and all status related to SBC connectivity are good and also Configured Voice routes with SBC for my user. I am facing one issue for calls from MS Teams to SBC and getting below error on MS teams Screen: *We couldn't connect the call.With your calling license, you can only call people within your organization. Talk to your IT admin to change your license.* What are the possible reasons for this error? Note: SBC to MS Teams Calls are working fine. I have configured below License in MS Teams account 1 ) Microsoft Teams Phone Standard 2 ) Microsoft Teams 3 ) Microsoft 365 Audio Conferencing Can Anyone tell me if I am missing any other license for Making outbound calls from MS Teams to SBC using Direct routing? Let me know if you require any further information related to MS Teams Configuration. Any suggestions would be appreciated. Many Thanks, Devang Dhandhalya -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 22 11:54:01 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 22 Apr 2024 17:24:01 +0530 Subject: [OpenSIPS-Users] call not landing in agents Message-ID: I've setup a call center using opensips. When I call to a number, the call should land in one of the call center agents. I've browser phone as agent1 and zoiper phone as agent2. The call is landing in agent2 after 45-50 mins of call establishment. But the call never lands in browser phone. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 22 13:44:37 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 22 Apr 2024 16:44:37 +0300 Subject: [OpenSIPS-Users] OpenSIPS 3.5 release dates Message-ID: The upcoming OpenSIPS 3.5 beta release is scheduled for *9th of May*, with just days before the OpenSIPS Summit in Valencia. It focuses on #IMS (IP Multimedia Subsystem), mainly on CSCF components - a lot of development was done in the area and still work-in-progress. And, as usual, it will be the star of the Summit 2024 😉 https://www.opensips.org/Development/Opensips-3-5-Planning https://www.opensips.org/events/Summit-2024Valencia/ Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmed.rehan at gmail.com Mon Apr 22 16:50:44 2024 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Mon, 22 Apr 2024 21:50:44 +0500 Subject: [OpenSIPS-Users] Ms Teams to SBC call not working In-Reply-To: References: Message-ID: Check your Voice Routing policy to the user you are calling from and also check the Route which is matching your dialed number has been assigned the SBC you have configured. On Mon, Apr 22, 2024 at 4:54 PM Devang Dhandhalya via Users < users at lists.opensips.org> wrote: > Hello All, > > I Configured Direct routing and all status related to SBC connectivity are > good and also Configured Voice routes with SBC for my user. > I am facing one issue for calls from MS Teams to SBC and getting below > error on MS teams Screen: > > > *We couldn't connect the call.With your calling license, you can only call > people within your organization. Talk to your IT admin to change your > license.* > What are the possible reasons for this error? > > Note: SBC to MS Teams Calls are working fine. > > I have configured below License in MS Teams account > > 1 ) Microsoft Teams Phone Standard > 2 ) Microsoft Teams > 3 ) Microsoft 365 Audio Conferencing > > Can Anyone tell me if I am missing any other license for Making outbound > calls from MS Teams to SBC using Direct routing? > Let me know if you require any further information related to MS Teams > Configuration. > Any suggestions would be appreciated. > > Many Thanks, > Devang Dhandhalya > > > *[image: https://www.ecosmob.com/opensips-summit/] > * > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From luisl at scarab.co.za Tue Apr 23 08:39:33 2024 From: luisl at scarab.co.za (Luis Leal) Date: Tue, 23 Apr 2024 08:39:33 +0000 Subject: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue Message-ID: <47660a2ed6764bc58081e8e2a54f1a86@scarab.co.za> Hi there, We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 and I was hoping someone would be able to shed some light on it. There are two symptoms: 1. Errors in the opensips log 2. SIPREC invite with correct SDP details (as per rtpengine log) but stream metadata missing from the XML metadata The errors in the log are as follows: Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:src_start_recording: could not start recording! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:tm_start_recording: cannot start recording! The curious part is that the above error happens before the 200 OK is received. The relevant SIP trace is: 12 21:33:50.208348 10.1.17.15 5060 10.249.224.9 54204 SIP/SDP 1125 Request: INVITE sip:1125 at 10.249.224.9:54204;ob | ...Snip... 21 21:33:53.262560 10.1.17.15 5060 10.1.17.12 5060 SIP/SDP 1073 Status: 200 OK (INVITE) | The SIPREC invite is still generated though but is missing stream details (participant details masked with +27XXXXXXXXX for privacy): Session Initiation Protocol (SIP as raw text) INVITE sip:10.1.17.24:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0 To: sip:10.1.17.24:5060 From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21 CSeq: 2 INVITE Call-ID: B2B.364.22430.1713814432.535273629 Max-Forwards: 70 Content-Length: 1995 User-Agent: OpenSIPS (3.4.4 (x86_64/linux)) Require: siprec Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 Contact: sip:10.1.17.15:5060;+sip.src --OSS-unique-boundary-42 Content-Type: application/sdp v=0 o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8 s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9 t=0 0 m=audio 31432 RTP/AVP 8 101 c=IN IP4 10.1.17.8 a=label:0 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1120210035 cname:060168be20ab122b a=sendonly a=rtcp:31433 m=audio 36760 RTP/AVP 8 101 c=IN IP4 10.1.17.8 a=label:1 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly a=rtcp:36761 a=ptime:20 --OSS-unique-boundary-42 Content-Type: application/rs-metadata+xml Content-Disposition: recording-session complete 2d409cb6-b066-4579-8c49-c6e6a7b9d600 +27XXXXXXXXX 2024-04-22T21:33:50+0200 2024-04-22T21:33:50+0200 2024-04-22T21:33:50+0200 --OSS-unique-boundary-42-- Is there a configuration item we're missing perhaps? Kind regards Luis Leal -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Tue Apr 23 09:09:23 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Tue, 23 Apr 2024 14:39:23 +0530 Subject: [OpenSIPS-Users] call not landing in agents In-Reply-To: References: Message-ID: Now the call is landing to an agents. But the music on hold continues without stopping. How to resolve this? On Mon, 22 Apr 2024 at 17:24, Prathibha B wrote: > I've setup a call center using opensips. When I call to a number, the call > should land in one of the call center agents. I've browser phone as agent1 > and zoiper phone as agent2. The call is landing in agent2 after 45-50 mins > of call establishment. But the call never lands in browser phone. > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From devang.dhandhalya at ecosmob.com Tue Apr 23 12:04:08 2024 From: devang.dhandhalya at ecosmob.com (Devang Dhandhalya) Date: Tue, 23 Apr 2024 17:34:08 +0530 Subject: [OpenSIPS-Users] Ms Teams to SBC call not working In-Reply-To: References: Message-ID: Hello Rehan, Thank you for the response I have configured this dialed pattern : ^+91(\d{10})$ and select My SBC which I have configured in Direct routing. I assigned a phone number and Direct routing for my user using Microsoft Teams admin center portal. I configure voice routing policy as well and not assign PSTN usage Record in it Also configured dialplan with dial pattern: ^+91(\d{10})$ and voice routing policy and dialplan is also assigned to my user Please let me know if I miss any further configuration in MS Teams. Regards, Devang Dhandhalya On Mon, Apr 22, 2024 at 10:20 PM Ahmed Rehan wrote: > Check your Voice Routing policy to the user you are calling from and also > check the Route which is matching your dialed number has been assigned the > SBC you have configured. > > On Mon, Apr 22, 2024 at 4:54 PM Devang Dhandhalya via Users < > users at lists.opensips.org> wrote: > >> Hello All, >> >> I Configured Direct routing and all status related to SBC connectivity >> are good and also Configured Voice routes with SBC for my user. >> I am facing one issue for calls from MS Teams to SBC and getting below >> error on MS teams Screen: >> >> >> *We couldn't connect the call.With your calling license, you can only >> call people within your organization. Talk to your IT admin to change your >> license.* >> What are the possible reasons for this error? >> >> Note: SBC to MS Teams Calls are working fine. >> >> I have configured below License in MS Teams account >> >> 1 ) Microsoft Teams Phone Standard >> 2 ) Microsoft Teams >> 3 ) Microsoft 365 Audio Conferencing >> >> Can Anyone tell me if I am missing any other license for Making outbound >> calls from MS Teams to SBC using Direct routing? >> Let me know if you require any further information related to MS Teams >> Configuration. >> Any suggestions would be appreciated. >> >> Many Thanks, >> Devang Dhandhalya >> >> >> *[image: https://www.ecosmob.com/opensips-summit/] >> * >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Regards > Ahmed Rehan > -- Regards, *Devang Dhandhalya* [image: Ecosmob Technologies Pvt. Ltd.] Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data -- * * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 24 08:27:54 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 24 Apr 2024 11:27:54 +0300 Subject: [OpenSIPS-Users] question on core statistics. In-Reply-To: References: Message-ID: Which queue are you referring at? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 22.04.2024 09:17, Johan De Clercq wrote: > Goed morning, > > How can I then increase opensips’s internal queue size? > > Best regards, Johan > > Verzonden vanuit Outlook voor iOS > ------------------------------------------------------------------------ > *Van:* Bogdan-Andrei Iancu > *Verzonden:* Thursday, April 18, 2024 4:29:41 PM > *Aan:* OpenSIPS users mailling list ; Johan > De Clercq > *Onderwerp:* Re: [OpenSIPS-Users] question on core statistics. > The `drop_requests` statistic is incremented when: > * the request is dropped by a pre-script callback (like B2B when there > is no script execution for certain messages) > * the stateless `forward()` core function failed to send out something. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 18.04.2024 17:19, Johan De Clercq wrote: >> No I don't. >> what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the >> default value of net.core.rmem_max and net.core.rmem_default. >> >> Op do 18 apr 2024 om 16:02 schreef Ben Newlin > >: >> >> Are you calling drop() anywhere in your script? >> >> https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13 >> >> >> Ben Newlin >> >> *From: *Users > > on behalf of Johan De >> Clercq > >> *Date: *Thursday, April 18, 2024 at 5:27 AM >> *To: *OpenSIPS users mailling list > > >> *Subject: *Re: [OpenSIPS-Users] question on core statistics. >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> ------------------------------------------------------------------------ >> >> would it make sense to recompile with other flags ? And how do I >> set them  (I don't find these of menuconfig's compile options)? >> >> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. >> >> Can somebody explain also what both flags mean. >> >> Op do 18 apr 2024 om 11:07 schreef Johan De Clercq >> >: >> >> would it make sense to recompile with other flags ? >> >> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535. >> >> Can somebody explain also what both flags mean. >> >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, >> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >> >> poll method support: poll, epoll, sigio_rt, select. >> >> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq >> >: >> >> Guys, >> >> I have an opensips instance running with 24 worker children. >> >> The worker load is very low. >> >> UDP queues are on 50 megs. >> >> when i query via the OS >> >> cat /proc/net/udp >> >>   sl  local_address rem_address   st tx_queue rx_queue tr >> tm->when retrnsmt   uid timeout inode ref pointer drops >> >>   590: 03231D0A:13C4 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 413684019 2 >> ffff880074820bc0 0 >> >>   591: 03231D0A:13C5 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 413766438 2 >> ffff880465e4a440 0 >> >>   592: 03231D0A:13C6 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 412035865 2 >> ffff8803e5a56b80 0 >> >>   934: 01231D0A:151C 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 26790 2 ffff88046c054840 0 >> >>   935: 0201FFEF:151D 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 26787 2 ffff88046c054bc0 0 >> >>   935: 01231D0A:151D 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 26791 2 ffff88046c0544c0 0 >> >>  1972: 00000000:D92A 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 15506 2 ffff88046dce5040 0 >> >>  5479: 00000000:E6DD 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 22811 2 ffff880465e4ab40 0 >> >> 12075: AA0914AC:00A1 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 20572 2 ffff88086d020800 0 >> >> 12075: 0100007F:00A1 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 20571 2 ffff88086d020b80 0 >> >> 13320: 00000000:857E 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000   100   0 17515 2 ffff8800368ac780 0 >> >> 15661: 00000000:CEA3 00000000:0000 07 00000000:00000000 >> 00:00000000 00000000     0   0 15505 2 ffff8800368acb00 0 >> >> => no drops >> >> what worries me is that there are drop requests and they  >> go up when I query via the mi interface >> >> opensipsctl fifo get_statistics drop_requests >> >> core:drop_requests:: 198107 >> >> opensipsctl fifo get_statistics drop_requests >> >> core:drop_requests:: 199157 >> >> opensipsctl_reg fifo get_statistics drop_requests >> >> core:drop_requests:: 204116 >> >> I don't see any memory issue, also the processload is low. >> >> so 3 questions: >> >> - what exactly is drop_request. >> >> - do I need to worry about this >> >> - how can I make them go lower. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 24 08:43:17 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 24 Apr 2024 11:43:17 +0300 Subject: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic In-Reply-To: References: Message-ID: Hi Julien, The term of AS is super abused when comes to what it should deliver. First of all you clearly need to define what should be the services/functionalities you need from an "AS" and to see what solutions are available to implement them. In most of the cases, OpenSIPS as proxy (dialog stateful) is able to provide most of them, without the need so any fancy external servlet or app - just using the OpenSIPS script. If there are good reasons to externalize the routing logic, you should consider more simple and flexible approach, take a look at this: https://blog.opensips.org/2023/03/22/api-driven-sip-user-agent-end-point-with-opensips-3-4/ Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 20.04.2024 11:50, julien.royannais at orange.com wrote: > > Hello everyone, > > I'm reaching out to get your opinion on using openSIPS with a business > Application Server (AS) that would handle the routing logic. > > At first glance, the SEAS module seems designed for this purpose, but > it doesn't appear to be a good fit as it seems highly coupled with a > Sip Servlet implementation using a specific protocol only supported > for WeSIP. > > It might be better to use REST interfaces, a 302 redirect-based > mechanism, or possibly another module. > > Thank you for your insights & advice! > > JR > > Orange Restricted > > > Orange Restricted > > ____________________________________________________________________________________________________________ > Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler > a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, > Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. > > This message and its attachments may contain confidential or privileged information that may be protected by law; > they should not be distributed, used or copied without authorisation. > If you have received this email in error, please notify the sender and delete this message and its attachments. > As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. > Thank you. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From luisl at scarab.co.za Wed Apr 24 09:16:19 2024 From: luisl at scarab.co.za (Luis Leal) Date: Wed, 24 Apr 2024 09:16:19 +0000 Subject: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue In-Reply-To: References: Message-ID: <177062d78dd641bcbf1a5a0fec48490e@scarab.co.za> Hi, I've done some exploring of the source code and have a hunch as to what's happening. The full trace is as follows where: 1. Endpoints = 10.1.17.12 / 10.249.224.9 2. SRS = 10.1.17.24 3. OpenSIPS = 10.1.17.15 12 21:33:50.208348 10.1.17.15 5060 10.249.224.9 54204 SIP/SDP 1125 Request: INVITE sip:1125 at 10.249.224.9:54204;ob | 13 21:33:50.235594 10.249.224.9 54204 10.1.17.15 5060 SIP 539 Status: 100 Trying | 14 21:33:50.235938 10.249.224.9 54204 10.1.17.15 5060 SIP 725 Status: 180 Ringing | 15 21:33:50.236090 10.1.17.15 5060 10.1.17.12 5060 SIP 641 Status: 180 Ringing | 20 21:33:53.262006 10.249.224.9 54204 10.1.17.15 5060 SIP/SDP 1171 Status: 200 OK (INVITE) | 21 21:33:53.262560 10.1.17.15 5060 10.1.17.12 5060 SIP/SDP 1073 Status: 200 OK (INVITE) | 22 21:33:53.263170 10.1.17.12 5060 10.1.17.15 5060 SIP 532 Request: ACK sip:1125 at 10.249.224.9:54204;ob | 24 21:33:53.263325 10.1.17.15 5060 10.1.17.24 5060 SIP/SDP/XML 1026 Request: INVITE sip:10.1.17.24:5060 | The errors logged: Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:src_start_recording: could not start recording! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:tm_start_recording: cannot start recording! It looks like either the callback tm_start_recording is being called too early on the 1XX packets or rtp_relay_copy_offer isn’t handling the unconfirmed session correctly? And perhaps the invite to the SRS shouldn’t be going out if there’s no RTP stream? I’m of course not an expert in the OpenSIPS architecture so I could be wrong. :) I’d appreciate it if someone more knowledgeable could confirm. Kind regards Luis Date: Tue, 23 Apr 2024 08:39:33 +0000 From: Luis Leal > To: "users at lists.opensips.org" > Subject: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue Message-ID: <47660a2ed6764bc58081e8e2a54f1a86 at scarab.co.za> Content-Type: text/plain; charset="utf-8" Hi there, We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 and I was hoping someone would be able to shed some light on it. There are two symptoms: 1. Errors in the opensips log 2. SIPREC invite with correct SDP details (as per rtpengine log) but stream metadata missing from the XML metadata The errors in the log are as follows: Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:src_start_recording: could not start recording! Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: ERROR:siprec:tm_start_recording: cannot start recording! The curious part is that the above error happens before the 200 OK is received. The relevant SIP trace is: 12 21:33:50.208348 10.1.17.15 5060 10.249.224.9 54204 SIP/SDP 1125 Request: INVITE sip:1125 at 10.249.224.9:54204;ob | ...Snip... 21 21:33:53.262560 10.1.17.15 5060 10.1.17.12 5060 SIP/SDP 1073 Status: 200 OK (INVITE) | The SIPREC invite is still generated though but is missing stream details (participant details masked with +27XXXXXXXXX for privacy): Session Initiation Protocol (SIP as raw text) INVITE sip:10.1.17.24:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0 To: sip:10.1.17.24:5060 From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21 CSeq: 2 INVITE Call-ID: B2B.364.22430.1713814432.535273629 Max-Forwards: 70 Content-Length: 1995 User-Agent: OpenSIPS (3.4.4 (x86_64/linux)) Require: siprec Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 Contact: sip:10.1.17.15:5060;+sip.src --OSS-unique-boundary-42 Content-Type: application/sdp v=0 o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8 s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9 t=0 0 m=audio 31432 RTP/AVP 8 101 c=IN IP4 10.1.17.8 a=label:0 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1120210035 cname:060168be20ab122b a=sendonly a=rtcp:31433 m=audio 36760 RTP/AVP 8 101 c=IN IP4 10.1.17.8 a=label:1 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly a=rtcp:36761 a=ptime:20 --OSS-unique-boundary-42 Content-Type: application/rs-metadata+xml Content-Disposition: recording-session complete 2d409cb6-b066-4579-8c49-c6e6a7b9d600 +27XXXXXXXXX 2024-04-22T21:33:50+0200 2024-04-22T21:33:50+0200 2024-04-22T21:33:50+0200 --OSS-unique-boundary-42-- Is there a configuration item we're missing perhaps? Kind regards Luis Leal -------------- next part -------------- An HTML attachment was scrubbed... URL: ------------------------------ Subject: Digest Footer _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ------------------------------ End of Users Digest, Vol 189, Issue 77 ************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 24 10:53:35 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 24 Apr 2024 13:53:35 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024 - This is our content Message-ID: <95cbf0bc-20f4-403b-a892-61d8cb5e80ab@opensips.org> OpenSIPS Summit, 14-17 May, 2024, Valencia, Spain * 2 days of conference on SIP, VoIP, RTC and Open Source * 1 day of demos * 1 day of advanced training * 1 cozy dinner event * 1 bold catamaran sea trip For such a great content let's boost the opportunity with the *SUMMIT-24-LABOR-DAY* /50% discount code/ between *1-5 of May* - this is a truly great deal ! Register now ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Wed Apr 24 14:21:24 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Wed, 24 Apr 2024 19:51:24 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: Now I've migrated to the latest version of opensips downloaded from github. Still call center module is not working for me. Error in syslog: ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key ERROR:b2b_logic:b2bl_restore_upper_info: Failed to parse b2b logic key ["User9"] ERROR:call_center:cc_db_restore_calls: Upper info not found for ["User9"] ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key ERROR:call_center:cc_db_insert_call: inserting new record in database ERROR:call_center:w_handle_call: Failed to insert call record in db ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method INVITE, On Sat, 20 Apr 2024 at 20:51, Prathibha B wrote: > when I run > > opensips-cli -x mi cc_list_calls > { > "Calls": [ > { > "id": "219.3", > "Ref": 2, > "State": "queued", > "Call Time": 187, > "Flow": "112" > }, > { > "id": "219.2", > "Ref": 2, > "State": "queued", > "Call Time": 187, > "Flow": "112" > }, > { > "id": "219.1", > "Ref": 2, > "State": "queued", > "Call Time": 758, > "Flow": "112" > }, > { > "id": "219.0", > "Ref": 2, > "State": "queued", > "Call Time": 758, > "Flow": "112" > } > ] > } > > opensips-cli -x mi cc_list_agents > { > "Agents": [ > { > "id": "101003", > "Ref": 0, > "Loged in": "YES", > "State": "wrapup", > "Wrapup-ends": 16 > }, > { > "id": "101002", > "Ref": 0, > "Loged in": "YES", > "State": "wrapup", > "Wrapup-ends": 27 > }, > { > "id": "101001", > "Ref": 0, > "Loged in": "YES", > "State": "wrapup", > "Wrapup-ends": 29 > } > ] > } > > On Sat, 20 Apr 2024 at 20:42, Prathibha B > wrote: > >> A non stop music is getting played but the call is not getting redirected >> to the call center agent. >> >> On Sat, 20 Apr 2024 at 14:17, Prathibha B >> wrote: >> >>> I am getting 401 Unauthorized error. How to resolve this issue? >>> >>> On Fri, 19 Apr 2024 at 19:13, Prathibha B >>> wrote: >>> >>>> Ok. >>>> >>>> Sent from Outlook for Android >>>> ------------------------------ >>>> *From:* Users on behalf of Brett >>>> Nemeroff >>>> *Sent:* Friday, April 19, 2024 7:08:51 PM >>>> *To:* OpenSIPS users mailling list >>>> *Subject:* Re: [OpenSIPS-Users] Call Center error >>>> >>>> I don’t know the details, but I’d guess that because you have no agents >>>> logged in your handle call isn’t properly rewriting the URI. It may be >>>> useful to check the return status and log it from cc_handle_call. That may >>>> give you some insight on at least this stage of the call processing. >>>> >>>> -Brett >>>> >>>> >>>> On Fri, Apr 19, 2024 at 6:09 AM Prathibha B >>>> wrote: >>>> >>>> or freeswitch as the media server. >>>> >>>> On Fri, 19 Apr 2024 at 14:45, Prathibha B >>>> wrote: >>>> >>>> Can someone in this group help me with setting up a call center using >>>> opensips as the proxy server and asterisk as media server? >>>> >>>> On Fri, 19 Apr 2024 at 10:04, Prathibha B >>>> wrote: >>>> >>>> I am getting the error "Cannot handle call" >>>> >>>> On Thu, 11 Apr 2024 at 16:16, Prathibha B >>>> wrote: >>>> >>>> I changed the message_queue uri in cc_flows table to sip:112 at bp.erss.in. >>>> Now getting the following errors in syslog: >>>> >>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>> change URI Contact >>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>> change URI Contact >>>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 >>>> was sent out >>>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>>> [B2B.394.162.1712831164.913185781] >>>> >>>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>>> wrote: >>>> >>>> when I run opensips-cli -x mi cc_list_agents >>>> { >>>> "Agents": [ >>>> { >>>> "id": "101002", >>>> "Ref": 0, >>>> "Loged in": "NO" >>>> }, >>>> { >>>> "id": "101001", >>>> "Ref": 0, >>>> "Loged in": "NO" >>>> } >>>> ] >>>> } >>>> >>>> I've logged in 101001 and 101002 in the browser. But the Loged in >>>> status is No for both users. >>>> >>>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>>> wrote: >>>> >>>> Cahnged it to $tU, Still getting error. >>>> >>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>> wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> Something wrong about that part for me : ($(tU) == "112") >>>> >>>> Should be : ($tU == "112") >>>> >>>> >>>> >>>> Regards >>>> >>>> >>>> >>>> *De : *Users au nom de Prathibha B < >>>> prathibhab.tvm at gmail.com> >>>> *Répondre à : *OpenSIPS users mailling list >>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>> *À : *OpenSIPS users mailling list >>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>> >>>> >>>> >>>> bp.erss.in - asterisk >>>> >>>> bp.erss.in:1443 - opensips >>>> >>>> >>>> >>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>> wrote: >>>> >>>> *I've created the entries in cc_agents and cc_flows table.* >>>> >>>> >>>> >>>> *cc_agents table* >>>> >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> | id | agentid | location | logstate | msrp_location >>>> | msrp_max_sessions | skills | wrapup_end_time | wrapup_time | >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>> | 4 | support | 0 | 0 | >>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>> | 4 | support | 0 | 0 | >>>> >>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>> >>>> >>>> >>>> *cc_flows table* >>>> >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>> | message_flow_id | >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> | 1 | support | 256 | support | NULL | 0 | >>>> 0 | 0 | 0 | >>>> 0 | | 112 at bp.erss.in | NULL | NULL >>>> | >>>> >>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>> >>>> >>>> >>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>> wrote: >>>> >>>> Getting the following error in call center module >>>> >>>> >>>> >>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID >>>> received) >>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>> >>>> >>>> >>>> Call center code in opensips.cfg: >>>> >>>> >>>> >>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>> if (!cc_handle_call("support")) { >>>> send_reply(403,"Cannot handle call"); >>>> exit; >>>> } >>>> >>>> } >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Regards, >>>> >>>> B.Prathibha >>>> >>>> _______________________________________________ Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From ionut.boangiu at itsyscom.com Thu Apr 25 09:38:43 2024 From: ionut.boangiu at itsyscom.com (Ionut Boangiu) Date: Thu, 25 Apr 2024 12:38:43 +0300 Subject: [OpenSIPS-Users] Error Initializing auth_aaa Module on Startup Message-ID: Hello, I've been working through the Diameter Authentication and Accounting tutorial (https://www.opensips.org/Documentation/Tutorials-Diameter-AAA) and I have some issues when loading the auth_aaa module on startup. Before posting, I had a look through the mailing list archives and did not find a solution to this issue. Sorry if I missed any. Debian 12 OpenSIPS Version: 3.5 nightly (also tried on 3.4 but encountered the same issue) freeDiameter version: 1.2.1 Error Logs: Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: INFO:auth_aaa:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: ERROR:aaa_diameter:dm_find: error in ((fd_dict_search(fd_g_config->cnf_dict, DICT_AVP, AVP_BY_NAME, map->name, &obj, 2))): -2 Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: ERROR:auth_aaa:mod_init: auth_aaa: can't get code for the Digest-Qop attribute (type 0) Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: ERROR:core:init_mod: failed to initialize module auth_aaa Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: ERROR:core:main: failed to initialize modules! What I did: - Installed freeDiameter and set up the configurations for both client and server (both sharing the same VM). - Verified that Capabilities and Watchdog exchanges happen successfully between the isolated client and server. - Created an opensips MySQL database for the diameter server using opensips-cli (currently empty). - Configured `opensips.cfg` using the residential setup as a base, modified the sockets, the module path, and loaded auth and diameter modules. - Included `aaa_proxy_authorize`, `proxy_challenge`, and `do_accounting` in the routing logic (without any purpose for now). - Started the freeDiameter server, then tried to start the opensips service and it failed with the error above. In the tutorial, no extensions were loaded in the client configuration. I thought initially, that I had to load the dict_sip.fdx extension for the client as well, as the Digest-Qop AVP is part of that, but that didn't make much sense after seeing in the debug logs that the other Digest-* AVPs were actually found. They are being loaded internally, right? I attached the logs, the opensips configuration and the freeDiameter configurations (for both server and client, although they should be identical to the tutorial with only the paths differing). Let me know if i missed anything. Any help is appreciated! I also have an unrelated question if you don't mind: is there a tool or method to convert a .c dict to .xml format, or is creating them myself/looking for them already in .xml format somewhere else the only option? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- Identity = "client.diameter.test"; Realm = "diameter.test"; Port = 3866; SecPort = 3867; No_SCTP; TLS_Cred = "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/cert.pem", "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/privkey.pem"; TLS_CA = "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/certchain.pem"; ConnectPeer = "server.diameter.test" { No_TLS; }; -------------- next part -------------- Apr 25 05:03:29 debian systemd[1]: Starting opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261) server... Apr 25 05:03:29 debian opensips[24398]: INFO:core:fix_poll_method: using epoll as the IO watch method (auto detected) Apr 25 05:03:29 debian opensips[24398]: Listening on Apr 25 05:03:29 debian opensips[24398]: udp: 127.0.0.1 [127.0.0.1]:5080 Apr 25 05:03:29 debian opensips[24398]: udp: 127.0.0.1 [127.0.0.1]:5060 Apr 25 05:03:29 debian opensips[24398]: udp: 192.168.122.42 [192.168.122.42]:5060 Apr 25 05:03:29 debian opensips[24398]: udp: 192.168.122.42 [192.168.122.42]:5080 Apr 25 05:03:29 debian opensips[24398]: Aliases: Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:core:main: version: opensips 3.5.0-dev (x86_64/linux) Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:core:main: using 64 MB of shared memory, allocator: Q_MALLOC_DBG Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:core:main: using 4 MB of private process memory, allocator: Q_MALLOC_DBG Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: WARNING:core:init_reactor_size: shrinking reactor size from 262144 (autodetected via rlimit) to 10485 (limited by memory of 10% from 4Mb) Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: WARNING:core:init_reactor_size: use 'open_files_limit' to enforce other limit or increase pkg memory Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:init_reactor_size: reactor size 10485 (using up to 0.40Mb of memory per process) Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:tm:mod_init: TM - initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:sl:mod_init: Initializing StateLess engine Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:signaling:mod_init: initializing module ... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:rr:mod_init: rr - initializing Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:maxfwd:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:sipmsgops:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:usrloc:ul_init_locks: locks array size 512 Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:registrar:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:acc:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:mod_init: initializing UDP-plain protocol Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:auth:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:aaa_diameter:dm_check_config: Diameter server support enabled Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:aaa_diameter:dm_check_config: Diameter server event E_DM_REQUEST not used in opensips script, auto-replying error code 3001 to any Diameter request Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:aaa_diameter:dm_init_minimal: initializing the Diameter object dictionary... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:evi_publish_event: Registered event Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:auth_aaa:mod_init: initializing... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: ERROR:aaa_diameter:dm_find: error in ((fd_dict_search(fd_g_config->cnf_dict, DICT_AVP, AVP_BY_NAME, map->name, &obj, 2))): -2 Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: ERROR:auth_aaa:mod_init: auth_aaa: can't get code for the Digest-Qop attribute (type 0) Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: ERROR:core:init_mod: failed to initialize module auth_aaa Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: ERROR:core:main: failed to initialize modules! Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: INFO:core:cleanup: cleanup Apr 25 05:03:29 debian opensips[24400]: 05:03:29 FATAL! Initiating freeDiameter shutdown sequence (0) Apr 25 05:03:29 debian opensips[24400]: 05:03:29 NOTI freeDiameterd framework is stopping... Apr 25 05:03:29 debian opensips[24400]: 05:03:29 NOTI Shutting down server sockets... Apr 25 05:03:29 debian opensips[24400]: 05:03:29 NOTI Sending terminate signal to all peer connections Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: NOTICE:core:main: Exiting.... Apr 25 05:03:29 debian /usr/sbin/opensips[24400]: CRITICAL:core:sig_usr: segfault in attendant (starter) process! Apr 25 05:03:29 debian opensips[24398]: INFO:core:daemonize: pre-daemon process exiting with -1 Apr 25 05:03:29 debian systemd[1]: opensips.service: Control process exited, code=exited, status=255/EXCEPTION Apr 25 05:03:29 debian systemd[1]: opensips.service: Failed with result 'exit-code'. Apr 25 05:03:29 debian systemd[1]: Failed to start opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261) server. Apr 25 05:03:29 debian systemd[1]: opensips.service: Scheduled restart job, restart counter is at 5. Apr 25 05:03:29 debian systemd[1]: Stopped opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261) server. Apr 25 05:03:29 debian systemd[1]: opensips.service: Start request repeated too quickly. Apr 25 05:03:29 debian systemd[1]: opensips.service: Failed with result 'exit-code'. Apr 25 05:03:29 debian systemd[1]: Failed to start opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261) server. -------------- next part -------------- Identity = "server.diameter.test"; Realm = "diameter.test"; Port = 3868; No_SCTP; TLS_Cred = "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/cert.pem", "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/privkey.pem"; TLS_CA = "/home/boangiu/src/freeDiameter/contrib/PKI/ca_script2/ca_data/my_diameter_ca/clients/*.diameter.test/certchain.pem"; LoadExtension = "/usr/lib/freeDiameter/dict_sip.fdx"; LoadExtension = "/home/boangiu/src/freeDiameter/fDbuild/extensions/app_opensips.fdx"; ConnectPeer = "client.diameter.test" { No_TLS; port = 3866; }; -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips.cfg Type: application/octet-stream Size: 5428 bytes Desc: not available URL: From liviu at opensips.org Thu Apr 25 16:49:47 2024 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 25 Apr 2024 19:49:47 +0300 Subject: [OpenSIPS-Users] Error Initializing auth_aaa Module on Startup In-Reply-To: References: Message-ID: Hello Ionuț, On 25.04.2024 12:38, Ionut Boangiu wrote: > Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: > ERROR:auth_aaa:mod_init: auth_aaa: can't get code for the Digest-Qop > attribute (type 0) Thanks for the report!  You actually ran into a bug there, and I just pushed a fix on `master` branch, see this commit .  It should become available along with tomorrow's nightly packages, if you don't build from source. About your C dev question: rather than looking for a tool or a method, why not look for an XML library that /you /like working with, first and foremost?  Once you've decided, just walk the dict, build the XML tree, then print it out as some kind of (char *) buffer using the library's functionality. Best regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS Summit 2024 May 14-17 Valencia |www.opensips.org/events -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Fri Apr 26 10:03:05 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Fri, 26 Apr 2024 15:33:05 +0530 Subject: [OpenSIPS-Users] opensips Message-ID: Does opensips call center module work for webrtc based video calls? -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From ionut.boangiu at itsyscom.com Mon Apr 29 08:23:49 2024 From: ionut.boangiu at itsyscom.com (Ionut Boangiu) Date: Mon, 29 Apr 2024 11:23:49 +0300 Subject: [OpenSIPS-Users] Error Initializing auth_aaa Module on Startup In-Reply-To: References: Message-ID: Thanks for the quick fix and for your answer to my other question! Can confirm that opensips is now starting properly after the update. I encountered some "programming bug" from do_accounting, but I'll open an issue on github for it. On Thu, Apr 25, 2024 at 7:49 PM Liviu Chircu wrote: > Hello Ionuț, > > On 25.04.2024 12:38, Ionut Boangiu wrote: > > Apr 25 05:03:29 debian /usr/sbin/opensips[24397]: ERROR:auth_aaa:mod_init: > auth_aaa: can't get code for the Digest-Qop attribute (type 0) > > Thanks for the report! You actually ran into a bug there, and I just > pushed a fix on `master` branch, see this commit > . > It should become available along with tomorrow's nightly packages, if you > don't build from source. > > About your C dev question: rather than looking for a tool or a method, why > not look for an XML library that *you *like working with, first and > foremost? Once you've decided, just walk the dict, build the XML tree, > then print it out as some kind of (char *) buffer using the library's > functionality. > > Best regards, > > -- > Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS Summit 2024 May 14-17 Valencia | www.opensips.org/events > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 29 09:00:38 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 29 Apr 2024 14:30:38 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: The call lands in agent2. But the call stops after 5 seconds in agent2. The MOH continues in the call initiator. On Wed, 24 Apr 2024 at 20:05, Prathibha B wrote: > config file attached. > > On Wed, 24 Apr 2024 at 19:51, Prathibha B > wrote: > >> Now I've migrated to the latest version of opensips downloaded from >> github. Still call center module is not working for me. Error in syslog: >> >> ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key >> ERROR:b2b_logic:b2bl_restore_upper_info: Failed to parse b2b logic key >> ["User9"] >> ERROR:call_center:cc_db_restore_calls: Upper info not found for ["User9"] >> ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key >> ERROR:call_center:cc_db_insert_call: inserting new record in database >> ERROR:call_center:w_handle_call: Failed to insert call record in db >> ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method >> INVITE, >> >> On Sat, 20 Apr 2024 at 20:51, Prathibha B >> wrote: >> >>> when I run >>> >>> opensips-cli -x mi cc_list_calls >>> { >>> "Calls": [ >>> { >>> "id": "219.3", >>> "Ref": 2, >>> "State": "queued", >>> "Call Time": 187, >>> "Flow": "112" >>> }, >>> { >>> "id": "219.2", >>> "Ref": 2, >>> "State": "queued", >>> "Call Time": 187, >>> "Flow": "112" >>> }, >>> { >>> "id": "219.1", >>> "Ref": 2, >>> "State": "queued", >>> "Call Time": 758, >>> "Flow": "112" >>> }, >>> { >>> "id": "219.0", >>> "Ref": 2, >>> "State": "queued", >>> "Call Time": 758, >>> "Flow": "112" >>> } >>> ] >>> } >>> >>> opensips-cli -x mi cc_list_agents >>> { >>> "Agents": [ >>> { >>> "id": "101003", >>> "Ref": 0, >>> "Loged in": "YES", >>> "State": "wrapup", >>> "Wrapup-ends": 16 >>> }, >>> { >>> "id": "101002", >>> "Ref": 0, >>> "Loged in": "YES", >>> "State": "wrapup", >>> "Wrapup-ends": 27 >>> }, >>> { >>> "id": "101001", >>> "Ref": 0, >>> "Loged in": "YES", >>> "State": "wrapup", >>> "Wrapup-ends": 29 >>> } >>> ] >>> } >>> >>> On Sat, 20 Apr 2024 at 20:42, Prathibha B >>> wrote: >>> >>>> A non stop music is getting played but the call is not getting >>>> redirected to the call center agent. >>>> >>>> On Sat, 20 Apr 2024 at 14:17, Prathibha B >>>> wrote: >>>> >>>>> I am getting 401 Unauthorized error. How to resolve this issue? >>>>> >>>>> On Fri, 19 Apr 2024 at 19:13, Prathibha B >>>>> wrote: >>>>> >>>>>> Ok. >>>>>> >>>>>> Sent from Outlook for Android >>>>>> ------------------------------ >>>>>> *From:* Users on behalf of Brett >>>>>> Nemeroff >>>>>> *Sent:* Friday, April 19, 2024 7:08:51 PM >>>>>> *To:* OpenSIPS users mailling list >>>>>> *Subject:* Re: [OpenSIPS-Users] Call Center error >>>>>> >>>>>> I don’t know the details, but I’d guess that because you have no >>>>>> agents logged in your handle call isn’t properly rewriting the URI. It may >>>>>> be useful to check the return status and log it from cc_handle_call. That >>>>>> may give you some insight on at least this stage of the call processing. >>>>>> >>>>>> -Brett >>>>>> >>>>>> >>>>>> On Fri, Apr 19, 2024 at 6:09 AM Prathibha B >>>>>> wrote: >>>>>> >>>>>> or freeswitch as the media server. >>>>>> >>>>>> On Fri, 19 Apr 2024 at 14:45, Prathibha B >>>>>> wrote: >>>>>> >>>>>> Can someone in this group help me with setting up a call center using >>>>>> opensips as the proxy server and asterisk as media server? >>>>>> >>>>>> On Fri, 19 Apr 2024 at 10:04, Prathibha B >>>>>> wrote: >>>>>> >>>>>> I am getting the error "Cannot handle call" >>>>>> >>>>>> On Thu, 11 Apr 2024 at 16:16, Prathibha B >>>>>> wrote: >>>>>> >>>>>> I changed the message_queue uri in cc_flows table to >>>>>> sip:112 at bp.erss.in. Now getting the following errors in syslog: >>>>>> >>>>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>>>> change URI Contact >>>>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>>>> change URI Contact >>>>>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 >>>>>> was sent out >>>>>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>>>>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>>>>> [B2B.394.162.1712831164.913185781] >>>>>> >>>>>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>>>>> wrote: >>>>>> >>>>>> when I run opensips-cli -x mi cc_list_agents >>>>>> { >>>>>> "Agents": [ >>>>>> { >>>>>> "id": "101002", >>>>>> "Ref": 0, >>>>>> "Loged in": "NO" >>>>>> }, >>>>>> { >>>>>> "id": "101001", >>>>>> "Ref": 0, >>>>>> "Loged in": "NO" >>>>>> } >>>>>> ] >>>>>> } >>>>>> >>>>>> I've logged in 101001 and 101002 in the browser. But the Loged in >>>>>> status is No for both users. >>>>>> >>>>>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>>>>> wrote: >>>>>> >>>>>> Cahnged it to $tU, Still getting error. >>>>>> >>>>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>>>> wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> >>>>>> >>>>>> Something wrong about that part for me : ($(tU) == "112") >>>>>> >>>>>> Should be : ($tU == "112") >>>>>> >>>>>> >>>>>> >>>>>> Regards >>>>>> >>>>>> >>>>>> >>>>>> *De : *Users au nom de Prathibha >>>>>> B >>>>>> *Répondre à : *OpenSIPS users mailling list >>>>> > >>>>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>>>> *À : *OpenSIPS users mailling list >>>>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>>>> >>>>>> >>>>>> >>>>>> bp.erss.in - asterisk >>>>>> >>>>>> bp.erss.in:1443 - opensips >>>>>> >>>>>> >>>>>> >>>>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>>>> wrote: >>>>>> >>>>>> *I've created the entries in cc_agents and cc_flows table.* >>>>>> >>>>>> >>>>>> >>>>>> *cc_agents table* >>>>>> >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> | id | agentid | location | logstate | >>>>>> msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time >>>>>> | >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>>>> | 4 | support | 0 | 0 | >>>>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>>>> | 4 | support | 0 | 0 | >>>>>> >>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>> >>>>>> >>>>>> >>>>>> *cc_flows table* >>>>>> >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>>>> | message_flow_id | >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> | 1 | support | 256 | support | NULL | 0 | >>>>>> 0 | 0 | 0 | >>>>>> 0 | | 112 at bp.erss.in | NULL | >>>>>> NULL | >>>>>> >>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>> >>>>>> >>>>>> >>>>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>>>> wrote: >>>>>> >>>>>> Getting the following error in call center module >>>>>> >>>>>> >>>>>> >>>>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty >>>>>> ID received) >>>>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>>>> >>>>>> >>>>>> >>>>>> Call center code in opensips.cfg: >>>>>> >>>>>> >>>>>> >>>>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>>>> if (!cc_handle_call("support")) { >>>>>> send_reply(403,"Cannot handle call"); >>>>>> exit; >>>>>> } >>>>>> >>>>>> } >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Regards, >>>>>> >>>>>> B.Prathibha >>>>>> >>>>>> _______________________________________________ Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> B.Prathibha >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From amel.guesmi at sofrecom.com Mon Apr 29 09:57:04 2024 From: amel.guesmi at sofrecom.com (amel.guesmi at sofrecom.com) Date: Mon, 29 Apr 2024 09:57:04 +0000 Subject: [OpenSIPS-Users] Tracer module integration with Opensips 3.4 Message-ID: <9aa3846765934357bfc70a8111dd2f8e@sofrecom.com> Hello Everyone, I need your support to add tracer module in order to store incoming/outgoing SIP messages in database. I already add some configs to my opensips.cfg file: ### Tracer ### loadmodule "tracer.so" modparam("tracer", "trace_on", 1) modparam("tracer", "trace_local_ip", "opensips:5060") modparam("tracer", "trace_id","[tid]uri=mysql://opensips:opensipsrw at ossdb/opensips;table=sip_trace;") .... if ( is_method("INVITE")) { record_route(); do_accounting("db|log", "cdr|missed", "acc"); trace($var(trace_id), "d", "sip|xlog", $var(user)); t_relay(); exit; } The error in Opensips logs is: ERROR:core:db_check_api: module db_mysql does not export db_use_table function 2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:get_db_struct: unable to bind database module 2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:parse_siptrace_id: Invalid parameters extracted!url ! table name ! 2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:parse_trace_id: failed to parse tracer uri [] 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:Traceback (last included file at the bottom): 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL: 0. /etc/opensips/opensips.cfg 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:core:yyerror: parse error in /etc/opensips/opensips.cfg:53:19-20: Parameter not found in module - can't set 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", "trace_on", 1) 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", "trace_local_ip", "opensips:5060") 2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", "trace_id","[tid]uri=mysql://opensips:opensipsrw at ossdb/opensips;table=sip_trace;") I think that the module should store the messages in sip_trace table but I didn't understand how to configure properly the trace_id with mysql module. Could you help me please ? Thank you. Best Regards, Amel on behalf of my colleague Chaker -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Apr 29 10:22:14 2024 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 29 Apr 2024 15:52:14 +0530 Subject: [OpenSIPS-Users] Call Center error In-Reply-To: References: Message-ID: If 1 agent rejects the call, will the call be forwarded to the next logged in agent? On Mon, 29 Apr 2024 at 14:30, Prathibha B wrote: > The call lands in agent2. But the call stops after 5 seconds in agent2. > The MOH continues in the call initiator. > > On Wed, 24 Apr 2024 at 20:05, Prathibha B > wrote: > >> config file attached. >> >> On Wed, 24 Apr 2024 at 19:51, Prathibha B >> wrote: >> >>> Now I've migrated to the latest version of opensips downloaded from >>> github. Still call center module is not working for me. Error in syslog: >>> >>> ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key >>> ERROR:b2b_logic:b2bl_restore_upper_info: Failed to parse b2b logic key >>> ["User9"] >>> ERROR:call_center:cc_db_restore_calls: Upper info not found for >>> ["User9"] >>> ERROR:b2b_logic:b2bl_parse_key: Wrong b2b logic key >>> ERROR:call_center:cc_db_insert_call: inserting new record in database >>> ERROR:call_center:w_handle_call: Failed to insert call record in db >>> ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method >>> INVITE, >>> >>> On Sat, 20 Apr 2024 at 20:51, Prathibha B >>> wrote: >>> >>>> when I run >>>> >>>> opensips-cli -x mi cc_list_calls >>>> { >>>> "Calls": [ >>>> { >>>> "id": "219.3", >>>> "Ref": 2, >>>> "State": "queued", >>>> "Call Time": 187, >>>> "Flow": "112" >>>> }, >>>> { >>>> "id": "219.2", >>>> "Ref": 2, >>>> "State": "queued", >>>> "Call Time": 187, >>>> "Flow": "112" >>>> }, >>>> { >>>> "id": "219.1", >>>> "Ref": 2, >>>> "State": "queued", >>>> "Call Time": 758, >>>> "Flow": "112" >>>> }, >>>> { >>>> "id": "219.0", >>>> "Ref": 2, >>>> "State": "queued", >>>> "Call Time": 758, >>>> "Flow": "112" >>>> } >>>> ] >>>> } >>>> >>>> opensips-cli -x mi cc_list_agents >>>> { >>>> "Agents": [ >>>> { >>>> "id": "101003", >>>> "Ref": 0, >>>> "Loged in": "YES", >>>> "State": "wrapup", >>>> "Wrapup-ends": 16 >>>> }, >>>> { >>>> "id": "101002", >>>> "Ref": 0, >>>> "Loged in": "YES", >>>> "State": "wrapup", >>>> "Wrapup-ends": 27 >>>> }, >>>> { >>>> "id": "101001", >>>> "Ref": 0, >>>> "Loged in": "YES", >>>> "State": "wrapup", >>>> "Wrapup-ends": 29 >>>> } >>>> ] >>>> } >>>> >>>> On Sat, 20 Apr 2024 at 20:42, Prathibha B >>>> wrote: >>>> >>>>> A non stop music is getting played but the call is not getting >>>>> redirected to the call center agent. >>>>> >>>>> On Sat, 20 Apr 2024 at 14:17, Prathibha B >>>>> wrote: >>>>> >>>>>> I am getting 401 Unauthorized error. How to resolve this issue? >>>>>> >>>>>> On Fri, 19 Apr 2024 at 19:13, Prathibha B >>>>>> wrote: >>>>>> >>>>>>> Ok. >>>>>>> >>>>>>> Sent from Outlook for Android >>>>>>> ------------------------------ >>>>>>> *From:* Users on behalf of Brett >>>>>>> Nemeroff >>>>>>> *Sent:* Friday, April 19, 2024 7:08:51 PM >>>>>>> *To:* OpenSIPS users mailling list >>>>>>> *Subject:* Re: [OpenSIPS-Users] Call Center error >>>>>>> >>>>>>> I don’t know the details, but I’d guess that because you have no >>>>>>> agents logged in your handle call isn’t properly rewriting the URI. It may >>>>>>> be useful to check the return status and log it from cc_handle_call. That >>>>>>> may give you some insight on at least this stage of the call processing. >>>>>>> >>>>>>> -Brett >>>>>>> >>>>>>> >>>>>>> On Fri, Apr 19, 2024 at 6:09 AM Prathibha B < >>>>>>> prathibhab.tvm at gmail.com> wrote: >>>>>>> >>>>>>> or freeswitch as the media server. >>>>>>> >>>>>>> On Fri, 19 Apr 2024 at 14:45, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> Can someone in this group help me with setting up a call center >>>>>>> using opensips as the proxy server and asterisk as media server? >>>>>>> >>>>>>> On Fri, 19 Apr 2024 at 10:04, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> I am getting the error "Cannot handle call" >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 16:16, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> I changed the message_queue uri in cc_flows table to >>>>>>> sip:112 at bp.erss.in. Now getting the following errors in syslog: >>>>>>> >>>>>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>>>>> change URI Contact >>>>>>> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to >>>>>>> change URI Contact >>>>>>> ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 >>>>>>> was sent out >>>>>>> ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm >>>>>>> ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, >>>>>>> [B2B.394.162.1712831164.913185781] >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 15:33, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> when I run opensips-cli -x mi cc_list_agents >>>>>>> { >>>>>>> "Agents": [ >>>>>>> { >>>>>>> "id": "101002", >>>>>>> "Ref": 0, >>>>>>> "Loged in": "NO" >>>>>>> }, >>>>>>> { >>>>>>> "id": "101001", >>>>>>> "Ref": 0, >>>>>>> "Loged in": "NO" >>>>>>> } >>>>>>> ] >>>>>>> } >>>>>>> >>>>>>> I've logged in 101001 and 101002 in the browser. But the Loged in >>>>>>> status is No for both users. >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 15:10, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> Cahnged it to $tU, Still getting error. >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent >>>>>>> wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> >>>>>>> >>>>>>> Something wrong about that part for me : ($(tU) == "112") >>>>>>> >>>>>>> Should be : ($tU == "112") >>>>>>> >>>>>>> >>>>>>> >>>>>>> Regards >>>>>>> >>>>>>> >>>>>>> >>>>>>> *De : *Users au nom de Prathibha >>>>>>> B >>>>>>> *Répondre à : *OpenSIPS users mailling list < >>>>>>> users at lists.opensips.org> >>>>>>> *Date : *jeudi 11 avril 2024 à 11:03 >>>>>>> *À : *OpenSIPS users mailling list >>>>>>> *Objet : *Re: [OpenSIPS-Users] Call Center error >>>>>>> >>>>>>> >>>>>>> >>>>>>> bp.erss.in - asterisk >>>>>>> >>>>>>> bp.erss.in:1443 - opensips >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 14:29, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> *I've created the entries in cc_agents and cc_flows table.* >>>>>>> >>>>>>> >>>>>>> >>>>>>> *cc_agents table* >>>>>>> >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> | id | agentid | location | logstate | >>>>>>> msrp_location | msrp_max_sessions | skills | wrapup_end_time | wrapup_time >>>>>>> | >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> | 8 | 101001 | sip:101001 at bp.erss.in:1443 | 0 | NULL >>>>>>> | 4 | support | 0 | 0 | >>>>>>> | 9 | 101002 | sip:101002 at bp.erss.in:1443 | 0 | NULL >>>>>>> | 4 | support | 0 | 0 | >>>>>>> >>>>>>> +----+---------+----------------------------+----------+---------------+-------------------+---------+-----------------+-------------+ >>>>>>> >>>>>>> >>>>>>> >>>>>>> *cc_flows table* >>>>>>> >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> | id | flowid | priority | skill | prependcid | max_wrapup_time | >>>>>>> dissuading_hangup | dissuading_onhold_th | dissuading_ewt_th | >>>>>>> dissuading_qsize_th | message_welcome | message_queue | message_dissuading >>>>>>> | message_flow_id | >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> | 1 | support | 256 | support | NULL | 0 | >>>>>>> 0 | 0 | 0 | >>>>>>> 0 | | 112 at bp.erss.in | NULL | >>>>>>> NULL | >>>>>>> >>>>>>> +----+---------+----------+---------+------------+-----------------+-------------------+----------------------+-------------------+---------------------+-----------------+----------------+--------------------+-----------------+ >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, 11 Apr 2024 at 14:27, Prathibha B >>>>>>> wrote: >>>>>>> >>>>>>> Getting the following error in call center module >>>>>>> >>>>>>> >>>>>>> >>>>>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty >>>>>>> ID received) >>>>>>> Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: >>>>>>> ERROR:call_center:w_handle_call: failed to set new destination for call >>>>>>> >>>>>>> >>>>>>> >>>>>>> Call center code in opensips.cfg: >>>>>>> >>>>>>> >>>>>>> >>>>>>> if (is_method("INVITE") and !has_totag() and ($(tU) == "112") ) { >>>>>>> if (!cc_handle_call("support")) { >>>>>>> send_reply(403,"Cannot handle call"); >>>>>>> exit; >>>>>>> } >>>>>>> >>>>>>> } >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> B.Prathibha >>>>>>> >>>>>>> _______________________________________________ Users mailing list >>>>>>> Users at lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users at lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> B.Prathibha >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users at lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> B.Prathibha >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> B.Prathibha >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> B.Prathibha >>>> >>> >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... 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